From 82e8d723e9e6698572098bf2976223d5069b34b5 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 4 Oct 2019 16:49:31 +0200 Subject: [PATCH 001/136] sound: Fix Kconfig indentation Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191004144931.3851-1-krzk@kernel.org Signed-off-by: Takashi Iwai --- sound/core/Kconfig | 28 ++++++++++---------- sound/drivers/Kconfig | 20 +++++++------- sound/firewire/Kconfig | 6 ++--- sound/isa/Kconfig | 18 ++++++------- sound/mips/Kconfig | 12 ++++----- sound/pci/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 14 +++++----- sound/soc/codecs/Kconfig | 32 +++++++++++------------ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 48 +++++++++++++++++----------------- sound/soc/pxa/Kconfig | 16 ++++++------ sound/soc/qcom/Kconfig | 20 +++++++------- sound/soc/samsung/Kconfig | 8 +++--- sound/soc/sof/imx/Kconfig | 12 ++++----- sound/soc/sof/intel/Kconfig | 22 ++++++++-------- sound/soc/xilinx/Kconfig | 20 +++++++------- sound/soc/zte/Kconfig | 12 ++++----- sound/usb/Kconfig | 30 ++++++++++----------- 18 files changed, 161 insertions(+), 161 deletions(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4ee79ad6ae22..4044c42d8595 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -72,11 +72,11 @@ config SND_PCM_OSS config SND_PCM_OSS_PLUGINS bool "OSS PCM (digital audio) API - Include plugin system" depends on SND_PCM_OSS - default y + default y help - If you disable this option, the ALSA's OSS PCM API will not - support conversion of channels, formats and rates. It will - behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. config SND_PCM_TIMER bool "PCM timer interface" if EXPERT @@ -128,13 +128,13 @@ config SND_SUPPORT_OLD_API or older). config SND_PROC_FS - bool "Sound Proc FS Support" if EXPERT - depends on PROC_FS - default y - help - Say 'N' to disable Sound proc FS, which may reduce code size about - 9KB on x86_64 platform. - If unsure say Y. + bool "Sound Proc FS Support" if EXPERT + depends on PROC_FS + default y + help + Say 'N' to disable Sound proc FS, which may reduce code size about + 9KB on x86_64 platform. + If unsure say Y. config SND_VERBOSE_PROCFS bool "Verbose procfs contents" @@ -142,8 +142,8 @@ config SND_VERBOSE_PROCFS default y help Say Y here to include code for verbose procfs contents (provides - useful information to developers when a problem occurs). On the - other side, it makes the ALSA subsystem larger. + useful information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. config SND_VERBOSE_PRINTK bool "Verbose printk" @@ -164,7 +164,7 @@ config SND_DEBUG_VERBOSE depends on SND_DEBUG help Say Y here to enable extra-verbose debugging messages. - + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. So, say Y only if you are ready to be annoyed. diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 09932cc98e9d..15d6d46acf9c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART - tristate - select SND_RAWMIDI + tristate + select SND_RAWMIDI config SND_OPL3_LIB tristate @@ -90,16 +90,16 @@ config SND_DUMMY will be called snd-dummy. config SND_ALOOP - tristate "Generic loopback driver (PCM)" - select SND_PCM - help - Say 'Y' or 'M' to include support for the PCM loopback device. + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. This module returns played samples back to the user space using the standard ALSA PCM device. The devices are routed 0->1 and - 1->0, where first number is the playback PCM device and second + 1->0, where first number is the playback PCM device and second number is the capture device. Module creates two PCM devices and configured number of substreams (see the pcm_substreams module - parameter). + parameter). The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% @@ -142,12 +142,12 @@ config SND_MTS64 select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with - additional SMPTE Timecode capabilities for the parallel port. + additional SMPTE Timecode capabilities for the parallel port. Say 'Y' to include support for this device. To compile this driver as a module, chose 'M' here: the module - will be called snd-mts64. + will be called snd-mts64. config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index b0a904cdb932..995c2cefc222 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -77,7 +77,7 @@ config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB select SND_HWDEP - help + help Say Y here to include support for FireWire devices based on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware: * Edirol FA-66/FA-101 @@ -111,8 +111,8 @@ config SND_BEBOB * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro - To compile this driver as a module, choose M here: the module - will be called snd-bebob. + To compile this driver as a module, choose M here: the module + will be called snd-bebob. config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..6ffa48dd5983 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -2,22 +2,22 @@ # ALSA ISA drivers config SND_WSS_LIB - tristate - select SND_PCM + tristate + select SND_PCM select SND_TIMER config SND_SB_COMMON - tristate + tristate config SND_SB8_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON config SND_SB16_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON menuconfig SND_ISA bool "ISA sound devices" diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 8a33402fd415..b497b803c834 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -14,15 +14,15 @@ config SND_SGI_O2 tristate "SGI O2 Audio" depends on SGI_IP32 select SND_PCM - help - Sound support for the SGI O2 Workstation. + help + Sound support for the SGI O2 Workstation. config SND_SGI_HAL2 - tristate "SGI HAL2 Audio" - depends on SGI_HAS_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 select SND_PCM - help - Sound support for the SGI Indy and Indigo2 Workstation. + help + Sound support for the SGI Indy and Indigo2 Workstation. endif # SND_MIPS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..93bc9bef7641 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -217,7 +217,7 @@ config SND_CMIPCI will be called snd-cmipci. config SND_OXYGEN_LIB - tristate + tristate config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2333efac758a..8039a8febefa 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -33,13 +33,13 @@ config SND_EP93XX_SOC_AC97 select SND_SOC_AC97_BUS config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. config SND_EP93XX_SOC_SIMONE tristate "SoC Audio support for Simplemachines Sim.One board" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89f8c5a..ef9d73b89623 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -257,16 +257,16 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - help - Normally ASoC codec drivers are only built if a machine driver which - uses them is also built since they are only usable with a machine - driver. Selecting this option will allow these drivers to be built - without an explicit machine driver for test and development purposes. + help + Normally ASoC codec drivers are only built if a machine driver which + uses them is also built since they are only usable with a machine + driver. Selecting this option will allow these drivers to be built + without an explicit machine driver for test and development purposes. Support for the bus types used to access the codecs to be built must be selected separately. - If unsure select "N". + If unsure select "N". config SND_SOC_88PM860X tristate @@ -570,8 +570,8 @@ config SND_SOC_CS42XX8_I2C # Cirrus Logic CS43130 HiFi DAC config SND_SOC_CS43130 - tristate "Cirrus Logic CS43130 CODEC" - depends on I2C + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" @@ -643,19 +643,19 @@ config SND_SOC_L3 tristate config SND_SOC_DA7210 - tristate + tristate config SND_SOC_DA7213 - tristate + tristate config SND_SOC_DA7218 tristate config SND_SOC_DA7219 - tristate + tristate config SND_SOC_DA732X - tristate + tristate config SND_SOC_DA9055 tristate @@ -717,7 +717,7 @@ config SND_SOC_INNO_RK3036 select REGMAP_MMIO config SND_SOC_ISABELLE - tristate + tristate config SND_SOC_LM49453 tristate @@ -988,7 +988,7 @@ config SND_SOC_RT5640 tristate config SND_SOC_RT5645 - tristate + tristate config SND_SOC_RT5651 tristate @@ -1220,7 +1220,7 @@ config SND_SOC_UDA134X tristate config SND_SOC_UDA1380 - tristate + tristate depends on I2C config SND_SOC_WCD9335 @@ -1348,7 +1348,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate config SND_SOC_WM8955 tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..597516c0c192 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -113,7 +113,7 @@ config SND_SOC_INTEL_SKYLAKE select SND_SOC_INTEL_CNL select SND_SOC_INTEL_CFL help - This is a backwards-compatible option to select all devices + This is a backwards-compatible option to select all devices supported by the Intel SST/Skylake driver. This option is no longer recommended and will be deprecated when the SOF driver is introduced. Distributions should explicitly diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7ab4a5f..882ff36a7c9c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -3,13 +3,13 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help - Intel ASoC Machine Drivers. If you have a Intel machine that - has an audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y - Note that the answer to this question doesn't directly affect the - kernel: saying N will just cause the configurator to skip all - the questions about Intel ASoC machine drivers. + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH @@ -114,11 +114,11 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 - help - This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5672 audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" @@ -311,20 +311,20 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + depends on SPI + select SND_SOC_RT5663 + select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI + select SND_SOC_MAX98927 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5663 + RT5514 + MAX98927. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" @@ -393,7 +393,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 213d4dab0346..295cfffa4646 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -190,14 +190,14 @@ config SND_PXA2XX_SOC_MAGICIAN HTC Magician. config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" @@ -205,7 +205,7 @@ config SND_PXA2XX_SOC_IMOTE2 select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help - Say Y if you want to add support for SoC audio on the + Say Y if you want to add support for SoC audio on the IMote 2. config SND_MMP_SOC_BROWNSTONE diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 60086858e920..6530d2462a9e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -3,8 +3,8 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help - Say Y or M if you want to add support to use audio devices - in Qualcomm Technologies SOC-based platforms. + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. config SND_SOC_LPASS_CPU tristate @@ -30,17 +30,17 @@ config SND_SOC_STORM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help - Say Y or M if you want add support for SoC audio on the - Qualcomm Technologies IPQ806X-based Storm board. + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help - Support for Qualcomm Technologies LPASS audio block in - APQ8016 SOC-based systems. - Say Y if you want to use audio devices on MI2S. + Support for Qualcomm Technologies LPASS audio block in + APQ8016 SOC-based systems. + Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON tristate @@ -93,9 +93,9 @@ config SND_SOC_MSM8996 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help - Support for Qualcomm Technologies LPASS audio block in - APQ8096 SoC-based systems. - Say Y if you want to use audio device on this SoCs + Support for Qualcomm Technologies LPASS audio block in + APQ8096 SoC-based systems. + Say Y if you want to use audio device on this SoCs config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..9304177de78a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -195,10 +195,10 @@ config SND_SOC_ODROID Say Y here to enable audio support for the Odroid XU3/XU4. config SND_SOC_ARNDALE_RT5631_ALC5631 - tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_RT5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on I2C + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..6315fba8ce71 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -5,9 +5,9 @@ config SND_SOC_SOF_IMX_TOPLEVEL depends on ARM64|| COMPILE_TEST depends on SND_SOC_SOF_OF help - This adds support for Sound Open Firmware for NXP i.MX platforms. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX platforms. + Say Y if you have such a device. + If unsure select "N". if SND_SOC_SOF_IMX_TOPLEVEL @@ -16,8 +16,8 @@ config SND_SOC_SOF_IMX8 depends on IMX_SCU depends on IMX_DSP help - This adds support for Sound Open Firmware for NXP i.MX8 platforms - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX8 platforms + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..18c41fdc47c8 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_PCI config SND_SOC_SOF_INTEL_HIFI_EP_IPC tristate help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_INTEL_ATOM_HIFI_EP @@ -217,31 +217,31 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the Tigerlake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tigerlake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_ELKHARTLAKE_SUPPORT bool "SOF support for ElkhartLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the ElkhartLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the ElkhartLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 69973179ef15..1d3586b68db7 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,15 +9,15 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" - help - Select this option to enable Xilinx audio formatter - support. This provides DMA platform device support for - audio functionality. + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" - help - Select this option to enable Xilinx SPDIF Audio. - This provides playback and capture of SPDIF audio in - AES format. + tristate "Audio support for the the Xilinx SPDIF" + help + Select this option to enable Xilinx SPDIF Audio. + This provides playback and capture of SPDIF audio in + AES format. diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a7842e4b791c..a23d4f13ca19 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -18,9 +18,9 @@ config ZX_I2S ZTE ZX I2S interface config ZX_TDM - tristate "ZTE ZX TDM Driver Support" - depends on COMMON_CLK - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to the - ZTE ZX TDM interface + tristate "ZTE ZX TDM Driver Support" + depends on COMMON_CLK + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to the + ZTE ZX TDM interface diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e2c53a0841da..059242f15d75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -107,24 +107,24 @@ config SND_USB_US122L will be called snd-usb-us122l. config SND_USB_6FIRE - tristate "TerraTec DMX 6Fire USB" - select FW_LOADER - select BITREVERSE - select SND_RAWMIDI - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for TerraTec 6fire DMX USB interface. + tristate "TerraTec DMX 6Fire USB" + select FW_LOADER + select BITREVERSE + select SND_RAWMIDI + select SND_PCM + select SND_VMASTER + help + Say Y here to include support for TerraTec 6fire DMX USB interface. - You will need firmware files in order to be able to use the device - after it has been coldstarted. An install script for the firmware - and further help can be found at - http://sixfireusb.sourceforge.net + You will need firmware files in order to be able to use the device + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net config SND_USB_HIFACE - tristate "M2Tech hiFace USB-SPDIF driver" - select SND_PCM - help + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help Select this option to include support for M2Tech hiFace USB-SPDIF interface. From d68c3123ba88dcd6692c90e698473d57856578e1 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:16 +0900 Subject: [PATCH 002/136] ALSA: firewire-lib: add a member into AMDTP domain for events per period In IEC 61883-6, it's called as 'event' what has presentation time represented by timestamp in CIP header. Although the ratio of the number of event against the number of data block is different depending on event data type represented by the specific field in CIP header, it's just one in the most cases supported by ALSA IEC 61883-1/6 engine. In 1394 OHCI specification, applications can schedule hardware IRQ by configuring descriptor with IRQ flag for packet against each isochronous cycle. For future commit, I use the hardware IRQ for isoc IT context to acknowledge the elapse of PCM period for both playback/capture directions on AMDTP streams in the same domain. This commit is a preparation for the above idea. This commit adds a member into AMDTP domain structure to record the number of PCM frames. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 4 ++++ sound/firewire/amdtp-stream.h | 10 ++++++++++ 2 files changed, 14 insertions(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index e50e28f77e74..838f695b20de 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1143,6 +1143,8 @@ int amdtp_domain_init(struct amdtp_domain *d) { INIT_LIST_HEAD(&d->streams); + d->events_per_period = 0; + return 0; } EXPORT_SYMBOL_GPL(amdtp_domain_init); @@ -1221,5 +1223,7 @@ void amdtp_domain_stop(struct amdtp_domain *d) amdtp_stream_stop(s); } + + d->events_per_period = 0; } EXPORT_SYMBOL_GPL(amdtp_domain_stop); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index bbbca964b9b4..d95a4ed15f20 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -272,6 +272,8 @@ static inline bool amdtp_stream_wait_callback(struct amdtp_stream *s, struct amdtp_domain { struct list_head streams; + + unsigned int events_per_period; }; int amdtp_domain_init(struct amdtp_domain *d); @@ -283,4 +285,12 @@ int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s, int amdtp_domain_start(struct amdtp_domain *d); void amdtp_domain_stop(struct amdtp_domain *d); +static inline int amdtp_domain_set_events_per_period(struct amdtp_domain *d, + unsigned int events_per_period) +{ + d->events_per_period = events_per_period; + + return 0; +} + #endif From 8737209fe4a251a7d1de8f77c126ce7b69d926e7 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:17 +0900 Subject: [PATCH 003/136] ALSA: bebob: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob.h | 3 ++- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 4 +++- sound/firewire/bebob/bebob_stream.c | 11 ++++++++++- 4 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 356d6ba60959..8738c1d8abf7 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -217,7 +217,8 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, enum snd_bebob_clock_type *src); int snd_bebob_stream_discover(struct snd_bebob *bebob); int snd_bebob_stream_init_duplex(struct snd_bebob *bebob); -int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate); +int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, + unsigned int frames_per_period); int snd_bebob_stream_start_duplex(struct snd_bebob *bebob); void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob); void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob); diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c index 4d8805fa8a00..e8f9edf50be5 100644 --- a/sound/firewire/bebob/bebob_midi.c +++ b/sound/firewire/bebob/bebob_midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) return err; mutex_lock(&bebob->mutex); - err = snd_bebob_stream_reserve_duplex(bebob, 0); + err = snd_bebob_stream_reserve_duplex(bebob, 0, 0); if (err >= 0) { ++bebob->substreams_counter; err = snd_bebob_stream_start_duplex(bebob); diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 0fb9eed46837..3b2227abf8a9 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -197,9 +197,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&bebob->mutex); - err = snd_bebob_stream_reserve_duplex(bebob, rate); + err = snd_bebob_stream_reserve_duplex(bebob, rate, + frames_per_period); if (err >= 0) ++bebob->substreams_counter; mutex_unlock(&bebob->mutex); diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 73fee991bd75..f1db3ddc3e00 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -554,7 +554,8 @@ static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, return cmp_connection_reserve(conn, amdtp_stream_get_max_payload(stream)); } -int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate) +int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, + unsigned int frames_per_period) { unsigned int curr_rate; int err; @@ -607,6 +608,14 @@ int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate) cmp_connection_release(&bebob->out_conn); return err; } + + err = amdtp_domain_set_events_per_period(&bebob->domain, + frames_per_period); + if (err < 0) { + cmp_connection_release(&bebob->out_conn); + cmp_connection_release(&bebob->in_conn); + return err; + } } return 0; From dd20e68a305630d7f808268d46709545c358a9b2 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:18 +0900 Subject: [PATCH 004/136] ALSA: fireworks: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks.h | 3 ++- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 4 +++- sound/firewire/fireworks/fireworks_stream.c | 11 ++++++++++- 4 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4cda297f8438..fc5f945a49ff 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -207,7 +207,8 @@ int snd_efw_command_get_sampling_rate(struct snd_efw *efw, unsigned int *rate); int snd_efw_command_set_sampling_rate(struct snd_efw *efw, unsigned int rate); int snd_efw_stream_init_duplex(struct snd_efw *efw); -int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate); +int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate, + unsigned int frames_per_period); int snd_efw_stream_start_duplex(struct snd_efw *efw); void snd_efw_stream_stop_duplex(struct snd_efw *efw); void snd_efw_stream_update_duplex(struct snd_efw *efw); diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c index a9f4a9630d15..e593f842ee8f 100644 --- a/sound/firewire/fireworks/fireworks_midi.c +++ b/sound/firewire/fireworks/fireworks_midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) goto end; mutex_lock(&efw->mutex); - err = snd_efw_stream_reserve_duplex(efw, 0); + err = snd_efw_stream_reserve_duplex(efw, 0, 0); if (err >= 0) { ++efw->substreams_counter; err = snd_efw_stream_start_duplex(efw); diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index a7025dccc754..35fc95283423 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -231,9 +231,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&efw->mutex); - err = snd_efw_stream_reserve_duplex(efw, rate); + err = snd_efw_stream_reserve_duplex(efw, rate, + frames_per_period); if (err >= 0) ++efw->substreams_counter; mutex_unlock(&efw->mutex); diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index f2de304d2f26..0787d5c3b01b 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -181,7 +181,8 @@ static int keep_resources(struct snd_efw *efw, struct amdtp_stream *stream, return cmp_connection_reserve(conn, amdtp_stream_get_max_payload(stream)); } -int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate) +int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate, + unsigned int frames_per_period) { unsigned int curr_rate; int err; @@ -228,6 +229,14 @@ int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate) cmp_connection_release(&efw->in_conn); return err; } + + err = amdtp_domain_set_events_per_period(&efw->domain, + frames_per_period); + if (err < 0) { + cmp_connection_release(&efw->in_conn); + cmp_connection_release(&efw->out_conn); + return err; + } } return 0; From 1d6a722c4a0146ac20fd0daa541a177279adcbb2 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:19 +0900 Subject: [PATCH 005/136] ALSA: oxfw: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-midi.c | 4 ++-- sound/firewire/oxfw/oxfw-pcm.c | 6 ++++-- sound/firewire/oxfw/oxfw-stream.c | 12 +++++++++++- sound/firewire/oxfw/oxfw.h | 3 ++- 4 files changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c index 9bdec08cb8ea..72db7a17d0ad 100644 --- a/sound/firewire/oxfw/oxfw-midi.c +++ b/sound/firewire/oxfw/oxfw-midi.c @@ -18,7 +18,7 @@ static int midi_capture_open(struct snd_rawmidi_substream *substream) mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, 0, 0); + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, 0, 0, 0); if (err >= 0) { ++oxfw->substreams_count; err = snd_oxfw_stream_start_duplex(oxfw); @@ -45,7 +45,7 @@ static int midi_playback_open(struct snd_rawmidi_substream *substream) mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, 0, 0); + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, 0, 0, 0); if (err >= 0) { ++oxfw->substreams_count; err = snd_oxfw_stream_start_duplex(oxfw); diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 7c6d1c277d4d..2d1b5038f7f6 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -221,10 +221,11 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int channels = params_channels(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&oxfw->mutex); err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, - rate, channels); + rate, channels, frames_per_period); if (err >= 0) ++oxfw->substreams_count; mutex_unlock(&oxfw->mutex); @@ -246,10 +247,11 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int channels = params_channels(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&oxfw->mutex); err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, - rate, channels); + rate, channels, frames_per_period); if (err >= 0) ++oxfw->substreams_count; mutex_unlock(&oxfw->mutex); diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 3c9a796b6526..7d2e88c5b73d 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -244,7 +244,8 @@ static int keep_resources(struct snd_oxfw *oxfw, struct amdtp_stream *stream) int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, struct amdtp_stream *stream, - unsigned int rate, unsigned int pcm_channels) + unsigned int rate, unsigned int pcm_channels, + unsigned int frames_per_period) { struct snd_oxfw_stream_formation formation; enum avc_general_plug_dir dir; @@ -305,6 +306,15 @@ int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, return err; } } + + err = amdtp_domain_set_events_per_period(&oxfw->domain, + frames_per_period); + if (err < 0) { + cmp_connection_release(&oxfw->in_conn); + if (oxfw->has_output) + cmp_connection_release(&oxfw->out_conn); + return err; + } } return 0; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index c9627b8c5d6e..422746ef2439 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -103,7 +103,8 @@ int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, int snd_oxfw_stream_init_duplex(struct snd_oxfw *oxfw); int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, struct amdtp_stream *stream, - unsigned int rate, unsigned int pcm_channels); + unsigned int rate, unsigned int pcm_channels, + unsigned int frames_per_period); int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_stop_duplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_destroy_duplex(struct snd_oxfw *oxfw); From 94c8101a2717a96457b474580e47142a0ef933c9 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:20 +0900 Subject: [PATCH 006/136] ALSA: dice: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-6-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 7 ++++++- sound/firewire/dice/dice-stream.c | 8 +++++++- sound/firewire/dice/dice.h | 3 ++- 4 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c index c9e19bddfc09..69c3c06bd7aa 100644 --- a/sound/firewire/dice/dice-midi.c +++ b/sound/firewire/dice/dice-midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) mutex_lock(&dice->mutex); - err = snd_dice_stream_reserve_duplex(dice, 0); + err = snd_dice_stream_reserve_duplex(dice, 0, 0); if (err >= 0) { ++dice->substreams_counter; err = snd_dice_stream_start_duplex(dice); diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 94a4dccfc381..570456a7751d 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -243,9 +243,14 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int events_per_period = params_period_size(hw_params); mutex_lock(&dice->mutex); - err = snd_dice_stream_reserve_duplex(dice, rate); + // For double_pcm_frame quirk. + if (rate > 96000) + events_per_period /= 2; + err = snd_dice_stream_reserve_duplex(dice, rate, + events_per_period); if (err >= 0) ++dice->substreams_counter; mutex_unlock(&dice->mutex); diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index f6a8627ae5a2..ef36bf588d11 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -278,7 +278,8 @@ static void finish_session(struct snd_dice *dice, struct reg_params *tx_params, snd_dice_transaction_clear_enable(dice); } -int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate) +int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate, + unsigned int events_per_period) { unsigned int curr_rate; int err; @@ -324,6 +325,11 @@ int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate) &rx_params); if (err < 0) goto error; + + err = amdtp_domain_set_events_per_period(&dice->domain, + events_per_period); + if (err < 0) + goto error; } return 0; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index fa6d74303f54..1f9e3502974e 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -210,7 +210,8 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice); void snd_dice_stream_stop_duplex(struct snd_dice *dice); int snd_dice_stream_init_duplex(struct snd_dice *dice); void snd_dice_stream_destroy_duplex(struct snd_dice *dice); -int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate); +int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate, + unsigned int events_per_period); void snd_dice_stream_update_duplex(struct snd_dice *dice); int snd_dice_stream_detect_current_formats(struct snd_dice *dice); From 18b7f18ff694983c69523e8136f9a256596edd59 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:21 +0900 Subject: [PATCH 007/136] ALSA: firewire-digi00x: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-7-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 4 +++- sound/firewire/digi00x/digi00x-stream.c | 11 ++++++++++- sound/firewire/digi00x/digi00x.h | 3 ++- 4 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c index 2b57ece89101..a407e795d8e7 100644 --- a/sound/firewire/digi00x/digi00x-midi.c +++ b/sound/firewire/digi00x/digi00x-midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) return err; mutex_lock(&dg00x->mutex); - err = snd_dg00x_stream_reserve_duplex(dg00x, 0); + err = snd_dg00x_stream_reserve_duplex(dg00x, 0, 0); if (err >= 0) { ++dg00x->substreams_counter; err = snd_dg00x_stream_start_duplex(dg00x); diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 18e561b26625..9ced309d61fa 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -167,9 +167,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&dg00x->mutex); - err = snd_dg00x_stream_reserve_duplex(dg00x, rate); + err = snd_dg00x_stream_reserve_duplex(dg00x, rate, + frames_per_period); if (err >= 0) ++dg00x->substreams_counter; mutex_unlock(&dg00x->mutex); diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index d6a92460060f..96d331e47b07 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -283,7 +283,8 @@ void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x) destroy_stream(dg00x, &dg00x->tx_stream); } -int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate) +int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate, + unsigned int frames_per_period) { unsigned int curr_rate; int err; @@ -315,6 +316,14 @@ int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate) fw_iso_resources_free(&dg00x->rx_resources); return err; } + + err = amdtp_domain_set_events_per_period(&dg00x->domain, + frames_per_period); + if (err < 0) { + fw_iso_resources_free(&dg00x->rx_resources); + fw_iso_resources_free(&dg00x->tx_resources); + return err; + } } return 0; diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 8041c65f2736..d93694282568 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -141,7 +141,8 @@ int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x, int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect); int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x); -int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate); +int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate, + unsigned int frames_per_period); int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x); void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x); void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x); From 262542ed93921c2632abb8a900daecff2b28057c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:22 +0900 Subject: [PATCH 008/136] ALSA: firewire-tascam: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-8-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-pcm.c | 4 +++- sound/firewire/tascam/tascam-stream.c | 11 ++++++++++- sound/firewire/tascam/tascam.h | 3 ++- 3 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 2377732caa52..cea26d5eff1a 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -99,9 +99,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&tscm->mutex); - err = snd_tscm_stream_reserve_duplex(tscm, rate); + err = snd_tscm_stream_reserve_duplex(tscm, rate, + frames_per_period); if (err >= 0) ++tscm->substreams_counter; mutex_unlock(&tscm->mutex); diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index adf69a520b80..8c04a0ad17d9 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -383,7 +383,8 @@ void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm) destroy_stream(tscm, &tscm->tx_stream); } -int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate) +int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate, + unsigned int frames_per_period) { unsigned int curr_rate; int err; @@ -413,6 +414,14 @@ int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate) fw_iso_resources_free(&tscm->tx_resources); return err; } + + err = amdtp_domain_set_events_per_period(&tscm->domain, + frames_per_period); + if (err < 0) { + fw_iso_resources_free(&tscm->tx_resources); + fw_iso_resources_free(&tscm->rx_resources); + return err; + } } return 0; diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 15bd335fa07f..32e72a25bf46 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -168,7 +168,8 @@ int snd_tscm_stream_get_clock(struct snd_tscm *tscm, int snd_tscm_stream_init_duplex(struct snd_tscm *tscm); void snd_tscm_stream_update_duplex(struct snd_tscm *tscm); void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm); -int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate); +int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate, + unsigned int frames_per_period); int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate); void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm); From 0d39cd0e449925a6221da48835ef962cd72f330f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:23 +0900 Subject: [PATCH 009/136] ALSA: firewire-motu: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-9-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 4 +++- sound/firewire/motu/motu-stream.c | 11 ++++++++++- sound/firewire/motu/motu.h | 3 ++- 4 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/firewire/motu/motu-midi.c b/sound/firewire/motu/motu-midi.c index 46a0035df31e..997dd6c8ec31 100644 --- a/sound/firewire/motu/motu-midi.c +++ b/sound/firewire/motu/motu-midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) mutex_lock(&motu->mutex); - err = snd_motu_stream_reserve_duplex(motu, 0); + err = snd_motu_stream_reserve_duplex(motu, 0, 0); if (err >= 0) { ++motu->substreams_counter; err = snd_motu_stream_start_duplex(motu); diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index aa2e584da6fe..9a54c562494b 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -202,9 +202,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&motu->mutex); - err = snd_motu_stream_reserve_duplex(motu, rate); + err = snd_motu_stream_reserve_duplex(motu, rate, + frames_per_period); if (err >= 0) ++motu->substreams_counter; mutex_unlock(&motu->mutex); diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 813e38e6a86e..52b7c375bb0b 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -133,7 +133,8 @@ int snd_motu_stream_cache_packet_formats(struct snd_motu *motu) return 0; } -int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate) +int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, + unsigned int frames_per_period) { unsigned int curr_rate; int err; @@ -171,6 +172,14 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate) fw_iso_resources_free(&motu->tx_resources); return err; } + + err = amdtp_domain_set_events_per_period(&motu->domain, + frames_per_period); + if (err < 0) { + fw_iso_resources_free(&motu->tx_resources); + fw_iso_resources_free(&motu->rx_resources); + return err; + } } return 0; diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 350ee2c16f4a..a419e6e7daed 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -154,7 +154,8 @@ void snd_motu_transaction_unregister(struct snd_motu *motu); int snd_motu_stream_init_duplex(struct snd_motu *motu); void snd_motu_stream_destroy_duplex(struct snd_motu *motu); int snd_motu_stream_cache_packet_formats(struct snd_motu *motu); -int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate); +int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, + unsigned int frames_per_period); int snd_motu_stream_start_duplex(struct snd_motu *motu); void snd_motu_stream_stop_duplex(struct snd_motu *motu); int snd_motu_stream_lock_try(struct snd_motu *motu); From 9d9ff58c2f45f2bf926f7ee7c70bcc01bad4c0e8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:24 +0900 Subject: [PATCH 010/136] ALSA: fireface: register the size of PCM period to AMDTP domain This commit is a preparation to share the size of PCM period between PCM substreams on AMDTP streams in the same domain. At this time, the size of PCM period in PCM substream which starts AMDTP streams in the same domain is recorded. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-10-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireface/ff-pcm.c | 3 ++- sound/firewire/fireface/ff-stream.c | 11 ++++++++++- sound/firewire/fireface/ff.h | 3 ++- 3 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index 9eab3ad283ce..4f2208202494 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -211,9 +211,10 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); + unsigned int frames_per_period = params_period_size(hw_params); mutex_lock(&ff->mutex); - err = snd_ff_stream_reserve_duplex(ff, rate); + err = snd_ff_stream_reserve_duplex(ff, rate, frames_per_period); if (err >= 0) ++ff->substreams_counter; mutex_unlock(&ff->mutex); diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index e8e6f9fd6433..d05e7d3055e1 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -106,7 +106,8 @@ void snd_ff_stream_destroy_duplex(struct snd_ff *ff) destroy_stream(ff, &ff->tx_stream); } -int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate) +int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate, + unsigned int frames_per_period) { unsigned int curr_rate; enum snd_ff_clock_src src; @@ -150,6 +151,14 @@ int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate) err = ff->spec->protocol->allocate_resources(ff, rate); if (err < 0) return err; + + err = amdtp_domain_set_events_per_period(&ff->domain, + frames_per_period); + if (err < 0) { + fw_iso_resources_free(&ff->tx_resources); + fw_iso_resources_free(&ff->rx_resources); + return err; + } } return 0; diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index b4c22ca6079e..970d4ae571ee 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -139,7 +139,8 @@ int snd_ff_stream_get_multiplier_mode(enum cip_sfc sfc, enum snd_ff_stream_mode *mode); int snd_ff_stream_init_duplex(struct snd_ff *ff); void snd_ff_stream_destroy_duplex(struct snd_ff *ff); -int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate); +int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate, + unsigned int frames_per_period); int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate); void snd_ff_stream_stop_duplex(struct snd_ff *ff); void snd_ff_stream_update_duplex(struct snd_ff *ff); From a80b29a41df097fbad6ab7106c0e6c972f1454b9 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:25 +0900 Subject: [PATCH 011/136] ALSA: bebob: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams and MIDI substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This idea has one issue because it's incremented for MIDI substreams as well. In current implementation, for a case that any MIDI substream run and a PCM substream is going to start, PCM application to start the PCM substream can decide hardware parameters by restart packet streaming. Just checking the substream count can brings regression. Now AMDTP domain structure has a member for the size of PCM period in PCM substream which starts AMDTP streams in domain. When the value has zero and the substream count is greater than 1, it means that any MIDI substream starts AMDTP streams in domain. Usage of the value can resolve the above issue. This commit replaces the check with the substream count and the value for the size of PCM period. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-11-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_pcm.c | 39 ++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 12 deletions(-) diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 3b2227abf8a9..516c9874f4a1 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -129,18 +129,17 @@ pcm_init_hw_params(struct snd_bebob *bebob, return err; } -static int -pcm_open(struct snd_pcm_substream *substream) +static int pcm_open(struct snd_pcm_substream *substream) { struct snd_bebob *bebob = substream->private_data; const struct snd_bebob_rate_spec *spec = bebob->spec->rate; - unsigned int sampling_rate; + struct amdtp_domain *d = &bebob->domain; enum snd_bebob_clock_type src; int err; err = snd_bebob_stream_lock_try(bebob); if (err < 0) - goto end; + return err; err = pcm_init_hw_params(bebob, substream); if (err < 0) @@ -150,15 +149,19 @@ pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; - /* - * When source of clock is internal or any PCM stream are running, - * the available sampling rate is limited at current sampling rate. - */ + mutex_lock(&bebob->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. if (src == SND_BEBOB_CLOCK_TYPE_EXTERNAL || - amdtp_stream_pcm_running(&bebob->tx_stream) || - amdtp_stream_pcm_running(&bebob->rx_stream)) { + (bebob->substreams_counter > 0 && d->events_per_period > 0)) { + unsigned int frames_per_period = d->events_per_period; + unsigned int sampling_rate; + err = spec->get(bebob, &sampling_rate); if (err < 0) { + mutex_unlock(&bebob->mutex); dev_err(&bebob->unit->device, "fail to get sampling rate: %d\n", err); goto err_locked; @@ -166,11 +169,23 @@ pcm_open(struct snd_pcm_substream *substream) substream->runtime->hw.rate_min = sampling_rate; substream->runtime->hw.rate_max = sampling_rate; + + if (frames_per_period > 0) { + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&bebob->mutex); + goto err_locked; + } + } } + mutex_unlock(&bebob->mutex); + snd_pcm_set_sync(substream); -end: - return err; + + return 0; err_locked: snd_bebob_stream_lock_release(bebob); return err; From 56faf928db4f68a76274d1675c11279357576226 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:26 +0900 Subject: [PATCH 012/136] ALSA: fireworks: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams and MIDI substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This idea has one issue because it's incremented for MIDI substreams as well. In current implementation, for a case that any MIDI substream run and a PCM substream is going to start, PCM application to start the PCM substream can decide hardware parameters by restart packet streaming. Just checking the substream count can brings regression. Now AMDTP domain structure has a member for the size of PCM period in PCM substream which starts AMDTP streams in domain. When the value has zero and the substream count is greater than 1, it means that any MIDI substream starts AMDTP streams in domain. Usage of the value can resolve the above issue. This commit replaces the check with the substream count and the value for the size of PCM period. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-12-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks_pcm.c | 39 +++++++++++++++++------- 1 file changed, 28 insertions(+), 11 deletions(-) diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 35fc95283423..314d1f8b8344 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -173,13 +173,13 @@ pcm_init_hw_params(struct snd_efw *efw, static int pcm_open(struct snd_pcm_substream *substream) { struct snd_efw *efw = substream->private_data; - unsigned int sampling_rate; + struct amdtp_domain *d = &efw->domain; enum snd_efw_clock_source clock_source; int err; err = snd_efw_stream_lock_try(efw); if (err < 0) - goto end; + return err; err = pcm_init_hw_params(efw, substream); if (err < 0) @@ -189,23 +189,40 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; - /* - * When source of clock is not internal or any PCM streams are running, - * available sampling rate is limited at current sampling rate. - */ + mutex_lock(&efw->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. if ((clock_source != SND_EFW_CLOCK_SOURCE_INTERNAL) || - amdtp_stream_pcm_running(&efw->tx_stream) || - amdtp_stream_pcm_running(&efw->rx_stream)) { + (efw->substreams_counter > 0 && d->events_per_period > 0)) { + unsigned int frames_per_period = d->events_per_period; + unsigned int sampling_rate; + err = snd_efw_command_get_sampling_rate(efw, &sampling_rate); - if (err < 0) + if (err < 0) { + mutex_unlock(&efw->mutex); goto err_locked; + } substream->runtime->hw.rate_min = sampling_rate; substream->runtime->hw.rate_max = sampling_rate; + + if (frames_per_period > 0) { + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&efw->mutex); + goto err_locked; + } + } } + mutex_unlock(&efw->mutex); + snd_pcm_set_sync(substream); -end: - return err; + + return 0; err_locked: snd_efw_stream_lock_release(efw); return err; From c0ede398b52654edb8b2a90be69dec2fb966f483 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:27 +0900 Subject: [PATCH 013/136] ALSA: oxfw: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams and MIDI substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This idea has one issue because it's incremented for MIDI substreams as well. In current implementation, for a case that any MIDI substream run and a PCM substream is going to start, PCM application to start the PCM substream can decide hardware parameters by restart packet streaming. Just checking the substream count can brings regression. Now AMDTP domain structure has a member for the size of PCM period in PCM substream which starts AMDTP streams in domain. When the value has zero and the substream count is greater than 1, it means that any MIDI substream starts AMDTP streams in domain. Usage of the value can resolve the above issue. This commit replaces the check with the substream count and the value for the size of PCM period. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-13-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-pcm.c | 39 ++++++++++++++++++++++++---------- 1 file changed, 28 insertions(+), 11 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 2d1b5038f7f6..f3e25898d270 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -170,30 +170,47 @@ static int limit_to_current_params(struct snd_pcm_substream *substream) static int pcm_open(struct snd_pcm_substream *substream) { struct snd_oxfw *oxfw = substream->private_data; + struct amdtp_domain *d = &oxfw->domain; int err; err = snd_oxfw_stream_lock_try(oxfw); if (err < 0) - goto end; + return err; err = init_hw_params(oxfw, substream); if (err < 0) goto err_locked; - /* - * When any PCM streams are already running, the available sampling - * rate is limited at current value. - */ - if (amdtp_stream_pcm_running(&oxfw->tx_stream) || - amdtp_stream_pcm_running(&oxfw->rx_stream)) { + mutex_lock(&oxfw->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. + if (oxfw->substreams_count > 0 && d->events_per_period > 0) { + unsigned int frames_per_period = d->events_per_period; + err = limit_to_current_params(substream); - if (err < 0) - goto end; + if (err < 0) { + mutex_unlock(&oxfw->mutex); + goto err_locked; + } + + if (frames_per_period > 0) { + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&oxfw->mutex); + goto err_locked; + } + } } + mutex_unlock(&oxfw->mutex); + snd_pcm_set_sync(substream); -end: - return err; + + return 0; err_locked: snd_oxfw_stream_lock_release(oxfw); return err; From a8fb2248028d49b250e8ad686bc7362b1ee4297f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:28 +0900 Subject: [PATCH 014/136] ALSA: dice: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams and MIDI substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This idea has one issue because it's incremented for MIDI substreams as well. In current implementation, for a case that any MIDI substream run and a PCM substream is going to start, PCM application to start the PCM substream can decide hardware parameters by restart packet streaming. Just checking the substream count can brings regression. Now AMDTP domain structure has a member for the size of PCM period in PCM substream which starts AMDTP streams in domain. When the value has zero and the substream count is greater than 1, it means that any MIDI substream starts AMDTP streams in domain. Usage of the value can resolve the above issue. This commit replaces the check with the substream count and the value for the size of PCM period. Dice hardware has a quirk called as 'Dual Wire'. For a case of higher sampling transmission frequency, this commit performs calculations between the number of PCM frames and the number of events in AMDTP stream. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-14-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-pcm.c | 43 ++++++++++++++++++++++++---------- 1 file changed, 31 insertions(+), 12 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 570456a7751d..813c9ffbeced 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -164,13 +164,14 @@ static int init_hw_info(struct snd_dice *dice, static int pcm_open(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; + struct amdtp_domain *d = &dice->domain; unsigned int source; bool internal; int err; err = snd_dice_stream_lock_try(dice); if (err < 0) - goto end; + return err; err = init_hw_info(dice, substream); if (err < 0) @@ -195,27 +196,45 @@ static int pcm_open(struct snd_pcm_substream *substream) break; } - /* - * When source of clock is not internal or any PCM streams are running, - * available sampling rate is limited at current sampling rate. - */ + mutex_lock(&dice->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. if (!internal || - amdtp_stream_pcm_running(&dice->tx_stream[0]) || - amdtp_stream_pcm_running(&dice->tx_stream[1]) || - amdtp_stream_pcm_running(&dice->rx_stream[0]) || - amdtp_stream_pcm_running(&dice->rx_stream[1])) { + (dice->substreams_counter > 0 && d->events_per_period > 0)) { + unsigned int frames_per_period = d->events_per_period; unsigned int rate; err = snd_dice_transaction_get_rate(dice, &rate); - if (err < 0) + if (err < 0) { + mutex_unlock(&dice->mutex); goto err_locked; + } + substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; + + if (frames_per_period > 0) { + // For double_pcm_frame quirk. + if (rate > 96000) + frames_per_period *= 2; + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&dice->mutex); + goto err_locked; + } + } } + mutex_unlock(&dice->mutex); + snd_pcm_set_sync(substream); -end: - return err; + + return 0; err_locked: snd_dice_stream_lock_release(dice); return err; From c36f8fcc584ce8b54916e4ebdab476288b245e90 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:29 +0900 Subject: [PATCH 015/136] ALSA: firewire-digi00x: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams and MIDI substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This idea has one issue because it's incremented for MIDI substreams as well. In current implementation, for a case that any MIDI substream run and a PCM substream is going to start, PCM application to start the PCM substream can decide hardware parameters by restart packet streaming. Just checking the substream count can brings regression. Now AMDTP domain structure has a member for the size of PCM period in PCM substream which starts AMDTP streams in domain. When the value has zero and the substream count is greater than 1, it means that any MIDI substream starts AMDTP streams in domain. Usage of the value can resolve the above issue. This commit replaces the check with the substream count and the value for the size of PCM period. I note that DOT AMDTP protocol has a quirk to use different transmission method of IEC 61883-6 for tx/rx streams; non-blocking in tx stream and blocking in rx stream. Although the difference of transmission method between tx/rx streams precisely brings different timing for a certain amount of events due to their different calculation for data blocks per packet, it's possible to approximate enough amount of events mostly has the same timing. Actually current ALSA IEC 61883-1/6 engine uses large amount of data blocks for each hardware IRQ (=16 packets). Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-15-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/digi00x/digi00x-pcm.c | 35 ++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 7 deletions(-) diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 9ced309d61fa..8dbfb3ff17f6 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -100,14 +100,14 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x, static int pcm_open(struct snd_pcm_substream *substream) { struct snd_dg00x *dg00x = substream->private_data; + struct amdtp_domain *d = &dg00x->domain; enum snd_dg00x_clock clock; bool detect; - unsigned int rate; int err; err = snd_dg00x_stream_lock_try(dg00x); if (err < 0) - goto end; + return err; err = pcm_init_hw_params(dg00x, substream); if (err < 0) @@ -127,19 +127,40 @@ static int pcm_open(struct snd_pcm_substream *substream) } } + mutex_lock(&dg00x->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. if ((clock != SND_DG00X_CLOCK_INTERNAL) || - amdtp_stream_pcm_running(&dg00x->rx_stream) || - amdtp_stream_pcm_running(&dg00x->tx_stream)) { + (dg00x->substreams_counter > 0 && d->events_per_period > 0)) { + unsigned int frames_per_period = d->events_per_period; + unsigned int rate; + err = snd_dg00x_stream_get_external_rate(dg00x, &rate); - if (err < 0) + if (err < 0) { + mutex_unlock(&dg00x->mutex); goto err_locked; + } substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; + + if (frames_per_period > 0) { + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&dg00x->mutex); + goto err_locked; + } + } } + mutex_unlock(&dg00x->mutex); + snd_pcm_set_sync(substream); -end: - return err; + + return 0; err_locked: snd_dg00x_stream_lock_release(dg00x); return err; From 6669a11d35bfa37f319da639594cdcbb164085f4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:30 +0900 Subject: [PATCH 016/136] ALSA: firewire-tascam: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This commit replaces the check with the substream count and the value for the size of PCM period. Unlike the other drivers in ALSA firewire stack, no MIDI substream is multiplexed into AMDTP stream. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-16-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-pcm.c | 34 +++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 8 deletions(-) diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index cea26d5eff1a..6cd3a420fbdf 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -43,13 +43,13 @@ static int pcm_init_hw_params(struct snd_tscm *tscm, static int pcm_open(struct snd_pcm_substream *substream) { struct snd_tscm *tscm = substream->private_data; + struct amdtp_domain *d = &tscm->domain; enum snd_tscm_clock clock; - unsigned int rate; int err; err = snd_tscm_stream_lock_try(tscm); if (err < 0) - goto end; + return err; err = pcm_init_hw_params(tscm, substream); if (err < 0) @@ -59,19 +59,37 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; - if (clock != SND_TSCM_CLOCK_INTERNAL || - amdtp_stream_pcm_running(&tscm->rx_stream) || - amdtp_stream_pcm_running(&tscm->tx_stream)) { + mutex_lock(&tscm->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. + if (clock != SND_TSCM_CLOCK_INTERNAL || tscm->substreams_counter > 0) { + unsigned int frames_per_period = d->events_per_period; + unsigned int rate; + err = snd_tscm_stream_get_rate(tscm, &rate); - if (err < 0) + if (err < 0) { + mutex_unlock(&tscm->mutex); goto err_locked; + } substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&tscm->mutex); + goto err_locked; + } } + mutex_unlock(&tscm->mutex); + snd_pcm_set_sync(substream); -end: - return err; + + return 0; err_locked: snd_tscm_stream_lock_release(tscm); return err; From 3fd80b2003882b6a328caff9e6b3a14bed61f27c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:31 +0900 Subject: [PATCH 017/136] ALSA: firewire-motu: use the same size of period for PCM substream in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams and MIDI substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This idea has one issue because it's incremented for MIDI substreams as well. In current implementation, for a case that any MIDI substream run and a PCM substream is going to start, PCM application to start the PCM substream can decide hardware parameters by restart packet streaming. Just checking the substream count can brings regression. Now AMDTP domain structure has a member for the size of PCM period in PCM substream which starts AMDTP streams in domain. When the value has zero and the substream count is greater than 1, it means that any MIDI substream starts AMDTP streams in domain. Usage of the value can resolve the above issue. This commit replaces the check with the substream count and the value for the size of PCM period. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-17-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-pcm.c | 29 +++++++++++++++++++++-------- 1 file changed, 21 insertions(+), 8 deletions(-) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 9a54c562494b..7bbf8b86a33d 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -134,8 +134,8 @@ static int pcm_open(struct snd_pcm_substream *substream) { struct snd_motu *motu = substream->private_data; const struct snd_motu_protocol *const protocol = motu->spec->protocol; + struct amdtp_domain *d = &motu->domain; enum snd_motu_clock_source src; - unsigned int rate; int err; err = snd_motu_stream_lock_try(motu); @@ -152,28 +152,41 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; - /* - * When source of clock is not internal or any PCM streams are running, - * available sampling rate is limited at current sampling rate. - */ err = protocol->get_clock_source(motu, &src); if (err < 0) goto err_locked; + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. if (src != SND_MOTU_CLOCK_SOURCE_INTERNAL || - amdtp_stream_pcm_running(&motu->tx_stream) || - amdtp_stream_pcm_running(&motu->rx_stream)) { + (motu->substreams_counter > 0 && d->events_per_period > 0)) { + unsigned int frames_per_period = d->events_per_period; + unsigned int rate; + err = protocol->get_clock_rate(motu, &rate); if (err < 0) goto err_locked; + substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; + + if (frames_per_period > 0) { + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&motu->mutex); + goto err_locked; + } + } } snd_pcm_set_sync(substream); mutex_unlock(&motu->mutex); - return err; + return 0; err_locked: mutex_unlock(&motu->mutex); snd_motu_stream_lock_release(motu); From 3aac32634172819ede79426c85e0a43ee7fcc3b3 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Mon, 7 Oct 2019 20:05:32 +0900 Subject: [PATCH 018/136] ALSA: fireface: use the same size of period for PCM substreams in AMDTP streams In current implementation, when opening a PCM substream, it's needed to check whether the opposite PCM substream runs. This is to assign effectual constraints (e.g. sampling rate) to opened PCM substream. The number of PCM substreams on AMDTP streams in domain is recorded in own structure. Usage of this count is an alternative of the above check. This is better because the count is incremented in pcm.hw_params earlier than pcm.trigger. This commit replaces the check with the substream count and the value for the size of PCM period. Unlike the other drivers in ALSA firewire stack, no MIDI substream is multiplexed into AMDTP stream. I note that Fireface AMDTP protocol has a quirk that tx stream includes blank isochronous cycle. The packet for blank cycle is equivalent to empty or NODATA packet in IEC 61883-6, thus the protocol is similar to blocking transmission method of IEC 61883-6. On the other hand, rx stream adopts non-blocking transmission method. Although the difference of transmission method between tx/rx streams precisely brings different timing for a certain amount of events due to their different calculation for data blocks per packet, it's possible to approximate enough amount of events mostly has the same timing. Actually current ALSA IEC 61883-1/6 engine uses large amount of data blocks for each hardware IRQ (=16 packets). Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191007110532.30270-18-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireface/ff-pcm.c | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index 4f2208202494..415bc9ccd1c7 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -139,6 +139,7 @@ static int pcm_init_hw_params(struct snd_ff *ff, static int pcm_open(struct snd_pcm_substream *substream) { struct snd_ff *ff = substream->private_data; + struct amdtp_domain *d = &ff->domain; unsigned int rate; enum snd_ff_clock_src src; int i, err; @@ -155,16 +156,21 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto release_lock; + mutex_lock(&ff->mutex); + + // When source of clock is not internal or any stream is reserved for + // transmission of PCM frames, the available sampling rate is limited + // at current one. if (src != SND_FF_CLOCK_SRC_INTERNAL) { for (i = 0; i < CIP_SFC_COUNT; ++i) { if (amdtp_rate_table[i] == rate) break; } - /* - * The unit is configured at sampling frequency which packet - * streaming engine can't support. - */ + + // The unit is configured at sampling frequency which packet + // streaming engine can't support. if (i >= CIP_SFC_COUNT) { + mutex_unlock(&ff->mutex); err = -EIO; goto release_lock; } @@ -172,14 +178,25 @@ static int pcm_open(struct snd_pcm_substream *substream) substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; } else { - if (amdtp_stream_pcm_running(&ff->rx_stream) || - amdtp_stream_pcm_running(&ff->tx_stream)) { + if (ff->substreams_counter > 0) { + unsigned int frames_per_period = d->events_per_period; + rate = amdtp_rate_table[ff->rx_stream.sfc]; substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + frames_per_period, frames_per_period); + if (err < 0) { + mutex_unlock(&ff->mutex); + goto release_lock; + } } } + mutex_unlock(&ff->mutex); + snd_pcm_set_sync(substream); return 0; From a0e023317e2d55c6b2fbf342c12d8a59797e1cff Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:13 +0900 Subject: [PATCH 019/136] ALSA: firewire-lib: use variable size of queue for isoc packets instead of fixed size The number of packets in packet buffer has been fixed number (=48) since first commit of ALSA IEC 61883-1/6 packet streaming engine. This commit allows the engine to use variable number of packets in the buffer. The size is calculated by a parameter in AMDTP domain structure surely to store the number of events in the packets of buffer. Although the value of parameter is expected to come from 'period size' parameter of PCM substream, at present 48 is still used. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 45 +++++++++++++-------- sound/firewire/amdtp-stream.h | 6 ++- sound/firewire/bebob/bebob_stream.c | 2 +- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/digi00x/digi00x-stream.c | 2 +- sound/firewire/fireface/ff-stream.c | 2 +- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/motu-stream.c | 2 +- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/tascam/tascam-stream.c | 2 +- 10 files changed, 41 insertions(+), 26 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 838f695b20de..1f38729c8e74 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -54,7 +54,6 @@ /* TODO: make these configurable */ #define INTERRUPT_INTERVAL 16 -#define QUEUE_LENGTH 48 // For iso header, tstamp and 2 CIP header. #define IR_CTX_HEADER_SIZE_CIP 16 @@ -451,7 +450,7 @@ static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params) goto end; } - if (++s->packet_index >= QUEUE_LENGTH) + if (++s->packet_index >= s->queue_size) s->packet_index = 0; end: return err; @@ -669,13 +668,14 @@ static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) } // Align to actual cycle count for the packet which is going to be scheduled. -// This module queued the same number of isochronous cycle as QUEUE_LENGTH to -// skip isochronous cycle, therefore it's OK to just increment the cycle by -// QUEUE_LENGTH for scheduled cycle. -static inline u32 compute_it_cycle(const __be32 ctx_header_tstamp) +// This module queued the same number of isochronous cycle as the size of queue +// to kip isochronous cycle, therefore it's OK to just increment the cycle by +// the size of queue for scheduled cycle. +static inline u32 compute_it_cycle(const __be32 ctx_header_tstamp, + unsigned int queue_size) { u32 cycle = compute_cycle_count(ctx_header_tstamp); - return increment_cycle_count(cycle, QUEUE_LENGTH); + return increment_cycle_count(cycle, queue_size); } static int generate_device_pkt_descs(struct amdtp_stream *s, @@ -689,7 +689,7 @@ static int generate_device_pkt_descs(struct amdtp_stream *s, for (i = 0; i < packets; ++i) { struct pkt_desc *desc = descs + i; - unsigned int index = (s->packet_index + i) % QUEUE_LENGTH; + unsigned int index = (s->packet_index + i) % s->queue_size; unsigned int cycle; unsigned int payload_length; unsigned int data_blocks; @@ -730,9 +730,9 @@ static void generate_ideal_pkt_descs(struct amdtp_stream *s, for (i = 0; i < packets; ++i) { struct pkt_desc *desc = descs + i; - unsigned int index = (s->packet_index + i) % QUEUE_LENGTH; + unsigned int index = (s->packet_index + i) % s->queue_size; - desc->cycle = compute_it_cycle(*ctx_header); + desc->cycle = compute_it_cycle(*ctx_header, s->queue_size); desc->syt = calculate_syt(s, desc->cycle); desc->data_blocks = calculate_data_blocks(s, desc->syt); @@ -779,12 +779,15 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, { struct amdtp_stream *s = private_data; const __be32 *ctx_header = header; - unsigned int packets = header_length / sizeof(*ctx_header); + unsigned int packets; int i; if (s->packet_index < 0) return; + // Calculate the number of packets in buffer and check XRUN. + packets = header_length / sizeof(*ctx_header); + generate_ideal_pkt_descs(s, s->pkt_descs, ctx_header, packets); process_ctx_payloads(s, s->pkt_descs, packets); @@ -828,7 +831,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, if (s->packet_index < 0) return; - // The number of packets in buffer. + // Calculate the number of packets in buffer and check XRUN. packets = header_length / s->ctx_data.tx.ctx_header_size; err = generate_device_pkt_descs(s, s->pkt_descs, ctx_header, packets); @@ -874,7 +877,7 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, context->callback.sc = in_stream_callback; } else { - cycle = compute_it_cycle(*ctx_header); + cycle = compute_it_cycle(*ctx_header, s->queue_size); context->callback.sc = out_stream_callback; } @@ -894,7 +897,8 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, * amdtp_stream_set_parameters() and it must be started before any PCM or MIDI * device can be started. */ -static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) +static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, + struct amdtp_domain *d) { static const struct { unsigned int data_block; @@ -908,6 +912,7 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) [CIP_SFC_88200] = { 0, 67 }, [CIP_SFC_176400] = { 0, 67 }, }; + unsigned int events_per_buffer = d->events_per_buffer; unsigned int ctx_header_size; unsigned int max_ctx_payload_size; enum dma_data_direction dir; @@ -953,7 +958,13 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) max_ctx_payload_size -= IT_PKT_HEADER_SIZE_CIP; } - err = iso_packets_buffer_init(&s->buffer, s->unit, QUEUE_LENGTH, + if (events_per_buffer == 0) + events_per_buffer = INTERRUPT_INTERVAL * 3; + + s->queue_size = DIV_ROUND_UP(CYCLES_PER_SECOND * events_per_buffer, + amdtp_rate_table[s->sfc]); + + err = iso_packets_buffer_init(&s->buffer, s->unit, s->queue_size, max_ctx_payload_size, dir); if (err < 0) goto err_unlock; @@ -981,7 +992,7 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) else s->tag = TAG_CIP; - s->pkt_descs = kcalloc(INTERRUPT_INTERVAL, sizeof(*s->pkt_descs), + s->pkt_descs = kcalloc(s->queue_size, sizeof(*s->pkt_descs), GFP_KERNEL); if (!s->pkt_descs) { err = -ENOMEM; @@ -1196,7 +1207,7 @@ int amdtp_domain_start(struct amdtp_domain *d) int err = 0; list_for_each_entry(s, &d->streams, list) { - err = amdtp_stream_start(s, s->channel, s->speed); + err = amdtp_stream_start(s, s->channel, s->speed, d); if (err < 0) break; } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index d95a4ed15f20..8d541727e437 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -117,6 +117,7 @@ struct amdtp_stream { /* For packet processing. */ struct fw_iso_context *context; struct iso_packets_buffer buffer; + unsigned int queue_size; int packet_index; struct pkt_desc *pkt_descs; int tag; @@ -274,6 +275,7 @@ struct amdtp_domain { struct list_head streams; unsigned int events_per_period; + unsigned int events_per_buffer; }; int amdtp_domain_init(struct amdtp_domain *d); @@ -286,9 +288,11 @@ int amdtp_domain_start(struct amdtp_domain *d); void amdtp_domain_stop(struct amdtp_domain *d); static inline int amdtp_domain_set_events_per_period(struct amdtp_domain *d, - unsigned int events_per_period) + unsigned int events_per_period, + unsigned int events_per_buffer) { d->events_per_period = events_per_period; + d->events_per_buffer = events_per_buffer; return 0; } diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index f1db3ddc3e00..e8e9eca6f116 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -610,7 +610,7 @@ int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, } err = amdtp_domain_set_events_per_period(&bebob->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { cmp_connection_release(&bebob->out_conn); cmp_connection_release(&bebob->in_conn); diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index ef36bf588d11..b4ef08bd7e8f 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -327,7 +327,7 @@ int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate, goto error; err = amdtp_domain_set_events_per_period(&dice->domain, - events_per_period); + events_per_period, 0); if (err < 0) goto error; } diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index 96d331e47b07..83659fc0ef25 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -318,7 +318,7 @@ int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate, } err = amdtp_domain_set_events_per_period(&dg00x->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { fw_iso_resources_free(&dg00x->rx_resources); fw_iso_resources_free(&dg00x->tx_resources); diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index d05e7d3055e1..cbe0e5087b05 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -153,7 +153,7 @@ int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate, return err; err = amdtp_domain_set_events_per_period(&ff->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { fw_iso_resources_free(&ff->tx_resources); fw_iso_resources_free(&ff->rx_resources); diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 0787d5c3b01b..e4d4dd08c60a 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -231,7 +231,7 @@ int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate, } err = amdtp_domain_set_events_per_period(&efw->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { cmp_connection_release(&efw->in_conn); cmp_connection_release(&efw->out_conn); diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 52b7c375bb0b..97245c670732 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -174,7 +174,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, } err = amdtp_domain_set_events_per_period(&motu->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { fw_iso_resources_free(&motu->tx_resources); fw_iso_resources_free(&motu->rx_resources); diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 7d2e88c5b73d..318de8217b3a 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -308,7 +308,7 @@ int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, } err = amdtp_domain_set_events_per_period(&oxfw->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { cmp_connection_release(&oxfw->in_conn); if (oxfw->has_output) diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 8c04a0ad17d9..1e4d44ed8bed 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -416,7 +416,7 @@ int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate, } err = amdtp_domain_set_events_per_period(&tscm->domain, - frames_per_period); + frames_per_period, 0); if (err < 0) { fw_iso_resources_free(&tscm->tx_resources); fw_iso_resources_free(&tscm->rx_resources); From 1fde7a447a7f423b893ac3fcac0ba65893924189 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:14 +0900 Subject: [PATCH 020/136] ALSA: bebob: share PCM buffer size for both direction This commit allows ALSA bebob driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob.h | 3 ++- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 12 +++++++++++- sound/firewire/bebob/bebob_stream.c | 5 +++-- 4 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index 8738c1d8abf7..d1ad9a8451bc 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -218,7 +218,8 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, int snd_bebob_stream_discover(struct snd_bebob *bebob); int snd_bebob_stream_init_duplex(struct snd_bebob *bebob); int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_bebob_stream_start_duplex(struct snd_bebob *bebob); void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob); void snd_bebob_stream_destroy_duplex(struct snd_bebob *bebob); diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c index e8f9edf50be5..6f597d03e7c1 100644 --- a/sound/firewire/bebob/bebob_midi.c +++ b/sound/firewire/bebob/bebob_midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) return err; mutex_lock(&bebob->mutex); - err = snd_bebob_stream_reserve_duplex(bebob, 0, 0); + err = snd_bebob_stream_reserve_duplex(bebob, 0, 0, 0); if (err >= 0) { ++bebob->substreams_counter; err = snd_bebob_stream_start_duplex(bebob); diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 516c9874f4a1..8b2e0ceffe82 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -157,6 +157,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (src == SND_BEBOB_CLOCK_TYPE_EXTERNAL || (bebob->substreams_counter > 0 && d->events_per_period > 0)) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; unsigned int sampling_rate; err = spec->get(bebob, &sampling_rate); @@ -178,6 +179,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&bebob->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&bebob->mutex); + goto err_locked; + } } } @@ -213,10 +222,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&bebob->mutex); err = snd_bebob_stream_reserve_duplex(bebob, rate, - frames_per_period); + frames_per_period, frames_per_buffer); if (err >= 0) ++bebob->substreams_counter; mutex_unlock(&bebob->mutex); diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index e8e9eca6f116..5e4a61458be2 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -555,7 +555,8 @@ static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, } int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { unsigned int curr_rate; int err; @@ -610,7 +611,7 @@ int snd_bebob_stream_reserve_duplex(struct snd_bebob *bebob, unsigned int rate, } err = amdtp_domain_set_events_per_period(&bebob->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { cmp_connection_release(&bebob->out_conn); cmp_connection_release(&bebob->in_conn); From 659c6af569a2bd08b3b3c1f5e445a58510de02c8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:15 +0900 Subject: [PATCH 021/136] ALSA: fireworks: share PCM buffer size for both direction This commit allows ALSA fireworks driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks.h | 3 ++- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +++++++++++- sound/firewire/fireworks/fireworks_stream.c | 5 +++-- 4 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index fc5f945a49ff..dda797209a27 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -208,7 +208,8 @@ int snd_efw_command_set_sampling_rate(struct snd_efw *efw, unsigned int rate); int snd_efw_stream_init_duplex(struct snd_efw *efw); int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_efw_stream_start_duplex(struct snd_efw *efw); void snd_efw_stream_stop_duplex(struct snd_efw *efw); void snd_efw_stream_update_duplex(struct snd_efw *efw); diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c index e593f842ee8f..84621e356848 100644 --- a/sound/firewire/fireworks/fireworks_midi.c +++ b/sound/firewire/fireworks/fireworks_midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) goto end; mutex_lock(&efw->mutex); - err = snd_efw_stream_reserve_duplex(efw, 0, 0); + err = snd_efw_stream_reserve_duplex(efw, 0, 0, 0); if (err >= 0) { ++efw->substreams_counter; err = snd_efw_stream_start_duplex(efw); diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 314d1f8b8344..abcc53dac8a5 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -197,6 +197,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if ((clock_source != SND_EFW_CLOCK_SOURCE_INTERNAL) || (efw->substreams_counter > 0 && d->events_per_period > 0)) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; unsigned int sampling_rate; err = snd_efw_command_get_sampling_rate(efw, &sampling_rate); @@ -215,6 +216,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&efw->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&efw->mutex); + goto err_locked; + } } } @@ -249,10 +258,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&efw->mutex); err = snd_efw_stream_reserve_duplex(efw, rate, - frames_per_period); + frames_per_period, frames_per_buffer); if (err >= 0) ++efw->substreams_counter; mutex_unlock(&efw->mutex); diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index e4d4dd08c60a..f35a33d4d4e6 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -182,7 +182,8 @@ static int keep_resources(struct snd_efw *efw, struct amdtp_stream *stream, } int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { unsigned int curr_rate; int err; @@ -231,7 +232,7 @@ int snd_efw_stream_reserve_duplex(struct snd_efw *efw, unsigned int rate, } err = amdtp_domain_set_events_per_period(&efw->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { cmp_connection_release(&efw->in_conn); cmp_connection_release(&efw->out_conn); From 3299d2a0f74c0774da0672d0bc54741da4bbda6e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:16 +0900 Subject: [PATCH 022/136] ALSA: oxfw: share PCM buffer size for both direction This commit allows ALSA oxfw driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-midi.c | 4 ++-- sound/firewire/oxfw/oxfw-pcm.c | 17 +++++++++++++++-- sound/firewire/oxfw/oxfw-stream.c | 5 +++-- sound/firewire/oxfw/oxfw.h | 3 ++- 4 files changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c index 72db7a17d0ad..775cba3f1f02 100644 --- a/sound/firewire/oxfw/oxfw-midi.c +++ b/sound/firewire/oxfw/oxfw-midi.c @@ -18,7 +18,7 @@ static int midi_capture_open(struct snd_rawmidi_substream *substream) mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, 0, 0, 0); + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, 0, 0, 0, 0); if (err >= 0) { ++oxfw->substreams_count; err = snd_oxfw_stream_start_duplex(oxfw); @@ -45,7 +45,7 @@ static int midi_playback_open(struct snd_rawmidi_substream *substream) mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, 0, 0, 0); + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, 0, 0, 0, 0); if (err >= 0) { ++oxfw->substreams_count; err = snd_oxfw_stream_start_duplex(oxfw); diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index f3e25898d270..ba586d1ac91d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -188,6 +188,7 @@ static int pcm_open(struct snd_pcm_substream *substream) // at current one. if (oxfw->substreams_count > 0 && d->events_per_period > 0) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; err = limit_to_current_params(substream); if (err < 0) { @@ -203,6 +204,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&oxfw->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&oxfw->mutex); + goto err_locked; + } } } @@ -239,10 +248,12 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(hw_params); unsigned int channels = params_channels(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&oxfw->mutex); err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, - rate, channels, frames_per_period); + rate, channels, frames_per_period, + frames_per_buffer); if (err >= 0) ++oxfw->substreams_count; mutex_unlock(&oxfw->mutex); @@ -265,10 +276,12 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(hw_params); unsigned int channels = params_channels(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&oxfw->mutex); err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, - rate, channels, frames_per_period); + rate, channels, frames_per_period, + frames_per_buffer); if (err >= 0) ++oxfw->substreams_count; mutex_unlock(&oxfw->mutex); diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 318de8217b3a..995e9c5bd84b 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -245,7 +245,8 @@ static int keep_resources(struct snd_oxfw *oxfw, struct amdtp_stream *stream) int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, struct amdtp_stream *stream, unsigned int rate, unsigned int pcm_channels, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { struct snd_oxfw_stream_formation formation; enum avc_general_plug_dir dir; @@ -308,7 +309,7 @@ int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, } err = amdtp_domain_set_events_per_period(&oxfw->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { cmp_connection_release(&oxfw->in_conn); if (oxfw->has_output) diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 422746ef2439..c30e537087b0 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -104,7 +104,8 @@ int snd_oxfw_stream_init_duplex(struct snd_oxfw *oxfw); int snd_oxfw_stream_reserve_duplex(struct snd_oxfw *oxfw, struct amdtp_stream *stream, unsigned int rate, unsigned int pcm_channels, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_stop_duplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_destroy_duplex(struct snd_oxfw *oxfw); From ecb40fd2c8afdb66da7e309b43c6dc90e419c2dc Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:17 +0900 Subject: [PATCH 023/136] ALSA: dice: share PCM buffer size for both direction This commit allows ALSA dice driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-6-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 20 +++++++++++++++++--- sound/firewire/dice/dice-stream.c | 5 +++-- sound/firewire/dice/dice.h | 3 ++- 4 files changed, 23 insertions(+), 7 deletions(-) diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c index 69c3c06bd7aa..4c2998034313 100644 --- a/sound/firewire/dice/dice-midi.c +++ b/sound/firewire/dice/dice-midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) mutex_lock(&dice->mutex); - err = snd_dice_stream_reserve_duplex(dice, 0, 0); + err = snd_dice_stream_reserve_duplex(dice, 0, 0, 0); if (err >= 0) { ++dice->substreams_counter; err = snd_dice_stream_start_duplex(dice); diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 813c9ffbeced..7c0c34c5bd47 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -204,6 +204,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (!internal || (dice->substreams_counter > 0 && d->events_per_period > 0)) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; unsigned int rate; err = snd_dice_transaction_get_rate(dice, &rate); @@ -217,8 +218,10 @@ static int pcm_open(struct snd_pcm_substream *substream) if (frames_per_period > 0) { // For double_pcm_frame quirk. - if (rate > 96000) + if (rate > 96000) { frames_per_period *= 2; + frames_per_buffer *= 2; + } err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, @@ -227,6 +230,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&dice->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&dice->mutex); + goto err_locked; + } } } @@ -263,13 +274,16 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int events_per_period = params_period_size(hw_params); + unsigned int events_per_buffer = params_buffer_size(hw_params); mutex_lock(&dice->mutex); // For double_pcm_frame quirk. - if (rate > 96000) + if (rate > 96000) { events_per_period /= 2; + events_per_buffer /= 2; + } err = snd_dice_stream_reserve_duplex(dice, rate, - events_per_period); + events_per_period, events_per_buffer); if (err >= 0) ++dice->substreams_counter; mutex_unlock(&dice->mutex); diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index b4ef08bd7e8f..0cff346e8052 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -279,7 +279,8 @@ static void finish_session(struct snd_dice *dice, struct reg_params *tx_params, } int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate, - unsigned int events_per_period) + unsigned int events_per_period, + unsigned int events_per_buffer) { unsigned int curr_rate; int err; @@ -327,7 +328,7 @@ int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate, goto error; err = amdtp_domain_set_events_per_period(&dice->domain, - events_per_period, 0); + events_per_period, events_per_buffer); if (err < 0) goto error; } diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 1f9e3502974e..16366773e22e 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -211,7 +211,8 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice); int snd_dice_stream_init_duplex(struct snd_dice *dice); void snd_dice_stream_destroy_duplex(struct snd_dice *dice); int snd_dice_stream_reserve_duplex(struct snd_dice *dice, unsigned int rate, - unsigned int events_per_period); + unsigned int events_per_period, + unsigned int events_per_buffer); void snd_dice_stream_update_duplex(struct snd_dice *dice); int snd_dice_stream_detect_current_formats(struct snd_dice *dice); From 76c4ecbe4b07c0d1b1aba31bf2a7f20cccbdbb4e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:18 +0900 Subject: [PATCH 024/136] ALSA: firewire-digi00x: share PCM buffer size for both direction This commit allows ALSA firewire-digi00x driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-7-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 12 +++++++++++- sound/firewire/digi00x/digi00x-stream.c | 5 +++-- sound/firewire/digi00x/digi00x.h | 3 ++- 4 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c index a407e795d8e7..68eb8c39afa6 100644 --- a/sound/firewire/digi00x/digi00x-midi.c +++ b/sound/firewire/digi00x/digi00x-midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) return err; mutex_lock(&dg00x->mutex); - err = snd_dg00x_stream_reserve_duplex(dg00x, 0, 0); + err = snd_dg00x_stream_reserve_duplex(dg00x, 0, 0, 0); if (err >= 0) { ++dg00x->substreams_counter; err = snd_dg00x_stream_start_duplex(dg00x); diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 8dbfb3ff17f6..c9a833dff20d 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -135,6 +135,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if ((clock != SND_DG00X_CLOCK_INTERNAL) || (dg00x->substreams_counter > 0 && d->events_per_period > 0)) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; unsigned int rate; err = snd_dg00x_stream_get_external_rate(dg00x, &rate); @@ -153,6 +154,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&dg00x->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&dg00x->mutex); + goto err_locked; + } } } @@ -189,10 +198,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&dg00x->mutex); err = snd_dg00x_stream_reserve_duplex(dg00x, rate, - frames_per_period); + frames_per_period, frames_per_buffer); if (err >= 0) ++dg00x->substreams_counter; mutex_unlock(&dg00x->mutex); diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index 83659fc0ef25..0c539188ba18 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -284,7 +284,8 @@ void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x) } int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { unsigned int curr_rate; int err; @@ -318,7 +319,7 @@ int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate, } err = amdtp_domain_set_events_per_period(&dg00x->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { fw_iso_resources_free(&dg00x->rx_resources); fw_iso_resources_free(&dg00x->tx_resources); diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index d93694282568..129de8edd5ea 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -142,7 +142,8 @@ int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect); int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x); int snd_dg00x_stream_reserve_duplex(struct snd_dg00x *dg00x, unsigned int rate, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x); void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x); void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x); From 128307d5f947d43a89e68dd8357879a0f111821d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:19 +0900 Subject: [PATCH 025/136] ALSA: firewire-tascam: share PCM buffer size for both direction This commit allows ALSA firewire-tascam driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-8-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/tascam/tascam-pcm.c | 12 +++++++++++- sound/firewire/tascam/tascam-stream.c | 5 +++-- sound/firewire/tascam/tascam.h | 3 ++- 3 files changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 6cd3a420fbdf..b18664fdf955 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -66,6 +66,7 @@ static int pcm_open(struct snd_pcm_substream *substream) // at current one. if (clock != SND_TSCM_CLOCK_INTERNAL || tscm->substreams_counter > 0) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; unsigned int rate; err = snd_tscm_stream_get_rate(tscm, &rate); @@ -83,6 +84,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&tscm->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&tscm->mutex); + goto err_locked; + } } mutex_unlock(&tscm->mutex); @@ -118,10 +127,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&tscm->mutex); err = snd_tscm_stream_reserve_duplex(tscm, rate, - frames_per_period); + frames_per_period, frames_per_buffer); if (err >= 0) ++tscm->substreams_counter; mutex_unlock(&tscm->mutex); diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 1e4d44ed8bed..a9b3b7eb6d21 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -384,7 +384,8 @@ void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm) } int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { unsigned int curr_rate; int err; @@ -416,7 +417,7 @@ int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate, } err = amdtp_domain_set_events_per_period(&tscm->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { fw_iso_resources_free(&tscm->tx_resources); fw_iso_resources_free(&tscm->rx_resources); diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 32e72a25bf46..78b7a08986a1 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -169,7 +169,8 @@ int snd_tscm_stream_init_duplex(struct snd_tscm *tscm); void snd_tscm_stream_update_duplex(struct snd_tscm *tscm); void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm); int snd_tscm_stream_reserve_duplex(struct snd_tscm *tscm, unsigned int rate, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate); void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm); From 0f5482e7875bc2d507e6d92fcc8de22ff75fbdda Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:20 +0900 Subject: [PATCH 026/136] ALSA: firewire-motu: share PCM buffer size for both direction This commit allows ALSA firewire-motu driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-9-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 12 +++++++++++- sound/firewire/motu/motu-stream.c | 5 +++-- sound/firewire/motu/motu.h | 3 ++- 4 files changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/firewire/motu/motu-midi.c b/sound/firewire/motu/motu-midi.c index 997dd6c8ec31..2365f7dfde26 100644 --- a/sound/firewire/motu/motu-midi.c +++ b/sound/firewire/motu/motu-midi.c @@ -17,7 +17,7 @@ static int midi_open(struct snd_rawmidi_substream *substream) mutex_lock(&motu->mutex); - err = snd_motu_stream_reserve_duplex(motu, 0, 0); + err = snd_motu_stream_reserve_duplex(motu, 0, 0, 0); if (err >= 0) { ++motu->substreams_counter; err = snd_motu_stream_start_duplex(motu); diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 7bbf8b86a33d..00e693da0cad 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -162,6 +162,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (src != SND_MOTU_CLOCK_SOURCE_INTERNAL || (motu->substreams_counter > 0 && d->events_per_period > 0)) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; unsigned int rate; err = protocol->get_clock_rate(motu, &rate); @@ -179,6 +180,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&motu->mutex); goto err_locked; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&motu->mutex); + goto err_locked; + } } } @@ -216,10 +225,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&motu->mutex); err = snd_motu_stream_reserve_duplex(motu, rate, - frames_per_period); + frames_per_period, frames_per_buffer); if (err >= 0) ++motu->substreams_counter; mutex_unlock(&motu->mutex); diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 97245c670732..9975770c9b1f 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -134,7 +134,8 @@ int snd_motu_stream_cache_packet_formats(struct snd_motu *motu) } int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { unsigned int curr_rate; int err; @@ -174,7 +175,7 @@ int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, } err = amdtp_domain_set_events_per_period(&motu->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { fw_iso_resources_free(&motu->tx_resources); fw_iso_resources_free(&motu->rx_resources); diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index a419e6e7daed..35ed8b6bb8e8 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -155,7 +155,8 @@ int snd_motu_stream_init_duplex(struct snd_motu *motu); void snd_motu_stream_destroy_duplex(struct snd_motu *motu); int snd_motu_stream_cache_packet_formats(struct snd_motu *motu); int snd_motu_stream_reserve_duplex(struct snd_motu *motu, unsigned int rate, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_motu_stream_start_duplex(struct snd_motu *motu); void snd_motu_stream_stop_duplex(struct snd_motu *motu); int snd_motu_stream_lock_try(struct snd_motu *motu); From 4de3eb062e2daf04646d23417a39b8843df4508b Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:21 +0900 Subject: [PATCH 027/136] ALSA: fireface: share PCM buffer size for both direction This commit allows ALSA fireface driver to share PCM buffer size for both capture and playback PCM substream. When AMDTP domain starts for one of the PCM substream, buffer size of the PCM substream is stores to AMDTP domain structure. Some AMDTP streams have already run with the buffer size when another PCM substream starts, therefore the PCM substream has a constraint to its buffer size. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-10-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireface/ff-pcm.c | 13 ++++++++++++- sound/firewire/fireface/ff-stream.c | 5 +++-- sound/firewire/fireface/ff.h | 3 ++- 3 files changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index 415bc9ccd1c7..005d959f8651 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -180,6 +180,7 @@ static int pcm_open(struct snd_pcm_substream *substream) } else { if (ff->substreams_counter > 0) { unsigned int frames_per_period = d->events_per_period; + unsigned int frames_per_buffer = d->events_per_buffer; rate = amdtp_rate_table[ff->rx_stream.sfc]; substream->runtime->hw.rate_min = rate; @@ -192,6 +193,14 @@ static int pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&ff->mutex); goto release_lock; } + + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + frames_per_buffer, frames_per_buffer); + if (err < 0) { + mutex_unlock(&ff->mutex); + goto release_lock; + } } } @@ -229,9 +238,11 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { unsigned int rate = params_rate(hw_params); unsigned int frames_per_period = params_period_size(hw_params); + unsigned int frames_per_buffer = params_buffer_size(hw_params); mutex_lock(&ff->mutex); - err = snd_ff_stream_reserve_duplex(ff, rate, frames_per_period); + err = snd_ff_stream_reserve_duplex(ff, rate, frames_per_period, + frames_per_buffer); if (err >= 0) ++ff->substreams_counter; mutex_unlock(&ff->mutex); diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index cbe0e5087b05..a13754f914e8 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -107,7 +107,8 @@ void snd_ff_stream_destroy_duplex(struct snd_ff *ff) } int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate, - unsigned int frames_per_period) + unsigned int frames_per_period, + unsigned int frames_per_buffer) { unsigned int curr_rate; enum snd_ff_clock_src src; @@ -153,7 +154,7 @@ int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate, return err; err = amdtp_domain_set_events_per_period(&ff->domain, - frames_per_period, 0); + frames_per_period, frames_per_buffer); if (err < 0) { fw_iso_resources_free(&ff->tx_resources); fw_iso_resources_free(&ff->rx_resources); diff --git a/sound/firewire/fireface/ff.h b/sound/firewire/fireface/ff.h index 970d4ae571ee..dc7a20f75983 100644 --- a/sound/firewire/fireface/ff.h +++ b/sound/firewire/fireface/ff.h @@ -140,7 +140,8 @@ int snd_ff_stream_get_multiplier_mode(enum cip_sfc sfc, int snd_ff_stream_init_duplex(struct snd_ff *ff); void snd_ff_stream_destroy_duplex(struct snd_ff *ff); int snd_ff_stream_reserve_duplex(struct snd_ff *ff, unsigned int rate, - unsigned int frames_per_period); + unsigned int frames_per_period, + unsigned int frames_per_buffer); int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate); void snd_ff_stream_stop_duplex(struct snd_ff *ff); void snd_ff_stream_update_duplex(struct snd_ff *ff); From e229853d505d7ab77e9b68b0ac91d19f48fe6d80 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:22 +0900 Subject: [PATCH 028/136] ALSA: firewire-lib: schedule hardware IRQ according to the size of PCM period ALSA IEC 61883-1/6 packet streaming engine controls 1394 OHCI controller to generate hardware IRQ for fixed number of isochronous packets (=16) since its first commit. This commit allow the engine to generate it for variable period according to the number of event to handle. For outgoing stream, internal calculator is used to check the accumulated events. For incoming stream, the number of data block in the packet of stream is used to check the accumulated events. When it's unavailable, fixed number of packet roughly calculated in advance is used instead of event counting. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-11-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 70 +++++++++++++++++++++++++++-------- sound/firewire/amdtp-stream.h | 3 ++ 2 files changed, 58 insertions(+), 15 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 1f38729c8e74..fd7c3e4d5374 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -52,9 +52,6 @@ #define CIP_FMT_AM 0x10 #define AMDTP_FDF_NO_DATA 0xff -/* TODO: make these configurable */ -#define INTERRUPT_INTERVAL 16 - // For iso header, tstamp and 2 CIP header. #define IR_CTX_HEADER_SIZE_CIP 16 // For iso header and tstamp. @@ -435,11 +432,12 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params) +static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params, + bool sched_irq) { int err; - params->interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); + params->interrupt = sched_irq; params->tag = s->tag; params->sy = 0; @@ -457,21 +455,21 @@ static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params) } static inline int queue_out_packet(struct amdtp_stream *s, - struct fw_iso_packet *params) + struct fw_iso_packet *params, bool sched_irq) { params->skip = !!(params->header_length == 0 && params->payload_length == 0); - return queue_packet(s, params); + return queue_packet(s, params, sched_irq); } static inline int queue_in_packet(struct amdtp_stream *s, - struct fw_iso_packet *params) + struct fw_iso_packet *params, bool sched_irq) { // Queue one packet for IR context. params->header_length = s->ctx_data.tx.ctx_header_size; params->payload_length = s->ctx_data.tx.max_ctx_payload_length; params->skip = false; - return queue_packet(s, params); + return queue_packet(s, params, sched_irq); } static void generate_cip_header(struct amdtp_stream *s, __be32 cip_header[2], @@ -779,6 +777,8 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, { struct amdtp_stream *s = private_data; const __be32 *ctx_header = header; + unsigned int events_per_period = s->events_per_period; + unsigned int event_count = s->event_count; unsigned int packets; int i; @@ -799,6 +799,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, struct fw_iso_packet params; __be32 header[IT_PKT_HEADER_SIZE_CIP / sizeof(__be32)]; } template = { {0}, {0} }; + bool sched_irq = false; if (s->ctx_data.rx.syt_override < 0) syt = desc->syt; @@ -809,12 +810,20 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, desc->data_blocks, desc->data_block_counter, syt, i); - if (queue_out_packet(s, &template.params) < 0) { + event_count += desc->data_blocks; + if (event_count >= events_per_period) { + event_count -= events_per_period; + sched_irq = true; + } + + if (queue_out_packet(s, &template.params, sched_irq) < 0) { cancel_stream(s); return; } } + s->event_count = event_count; + fw_iso_context_queue_flush(s->context); } @@ -823,8 +832,10 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int packets; __be32 *ctx_header = header; + unsigned int events_per_period = s->events_per_period; + unsigned int event_count = s->event_count; + unsigned int packets; int i; int err; @@ -845,14 +856,29 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, } for (i = 0; i < packets; ++i) { + const struct pkt_desc *desc = s->pkt_descs + i; struct fw_iso_packet params = {0}; + bool sched_irq = false; - if (queue_in_packet(s, ¶ms) < 0) { + if (err >= 0) { + event_count += desc->data_blocks; + if (event_count >= events_per_period) { + event_count -= events_per_period; + sched_irq = true; + } + } else { + sched_irq = + !((s->packet_index + 1) % s->idle_irq_interval); + } + + if (queue_in_packet(s, ¶ms, sched_irq) < 0) { cancel_stream(s); return; } } + s->event_count = event_count; + fw_iso_context_queue_flush(s->context); } @@ -913,6 +939,7 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, [CIP_SFC_176400] = { 0, 67 }, }; unsigned int events_per_buffer = d->events_per_buffer; + unsigned int events_per_period = d->events_per_period; unsigned int ctx_header_size; unsigned int max_ctx_payload_size; enum dma_data_direction dir; @@ -958,11 +985,21 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, max_ctx_payload_size -= IT_PKT_HEADER_SIZE_CIP; } + // This is a case that AMDTP streams in domain run just for MIDI + // substream. Use the number of events equivalent to 10 msec as + // interval of hardware IRQ. + if (events_per_period == 0) + events_per_period = amdtp_rate_table[s->sfc] / 100; if (events_per_buffer == 0) - events_per_buffer = INTERRUPT_INTERVAL * 3; + events_per_buffer = events_per_period * 3; + s->idle_irq_interval = + DIV_ROUND_UP(CYCLES_PER_SECOND * events_per_period, + amdtp_rate_table[s->sfc]); s->queue_size = DIV_ROUND_UP(CYCLES_PER_SECOND * events_per_buffer, amdtp_rate_table[s->sfc]); + s->events_per_period = events_per_period; + s->event_count = 0; err = iso_packets_buffer_init(&s->buffer, s->unit, s->queue_size, max_ctx_payload_size, dir); @@ -1002,12 +1039,15 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, s->packet_index = 0; do { struct fw_iso_packet params; + bool sched_irq; + + sched_irq = !((s->packet_index + 1) % s->idle_irq_interval); if (s->direction == AMDTP_IN_STREAM) { - err = queue_in_packet(s, ¶ms); + err = queue_in_packet(s, ¶ms, sched_irq); } else { params.header_length = 0; params.payload_length = 0; - err = queue_out_packet(s, ¶ms); + err = queue_out_packet(s, ¶ms, sched_irq); } if (err < 0) goto err_pkt_descs; diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8d541727e437..344818e936df 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -145,6 +145,9 @@ struct amdtp_stream { int syt_override; } rx; } ctx_data; + unsigned int event_count; + unsigned int events_per_period; + unsigned int idle_irq_interval; /* For CIP headers. */ unsigned int source_node_id_field; From 99921ec67dcdaeb5a5cc503f23ea2b5b6ff380ae Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:23 +0900 Subject: [PATCH 029/136] ALSA: firewire-lib: tune the maximum available size of PCM period Linux driver for 1394 OHCI controller voluntarily flushes isoc context when total size of accumulated context header reached PAGE_SIZE. This kicks tasklet for the isoc context. This is inconvenient to process runtime of PCM substream. This commit adds a restriction of the maximum size of PCM period to avoid this behaviour. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-12-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 29 ++++++++++++++++++----------- 1 file changed, 18 insertions(+), 11 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index fd7c3e4d5374..8a7818130382 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -176,6 +176,8 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime) { struct snd_pcm_hardware *hw = &runtime->hw; + unsigned int ctx_header_size; + unsigned int maximum_usec_per_period; int err; hw->info = SNDRV_PCM_INFO_BATCH | @@ -196,19 +198,24 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, hw->period_bytes_max = hw->period_bytes_min * 2048; hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; - /* - * Currently firewire-lib processes 16 packets in one software - * interrupt callback. This equals to 2msec but actually the - * interval of the interrupts has a jitter. - * Additionally, even if adding a constraint to fit period size to - * 2msec, actual calculated frames per period doesn't equal to 2msec, - * depending on sampling rate. - * Anyway, the interval to call snd_pcm_period_elapsed() cannot 2msec. - * Here let us use 5msec for safe period interrupt. - */ + // Linux driver for 1394 OHCI controller voluntarily flushes isoc + // context when total size of accumulated context header reaches + // PAGE_SIZE. This kicks tasklet for the isoc context and brings + // callback in the middle of scheduled interrupts. + // Although AMDTP streams in the same domain use the same events per + // IRQ, use the largest size of context header between IT/IR contexts. + // Here, use the value of context header in IR context is for both + // contexts. + if (!(s->flags & CIP_NO_HEADER)) + ctx_header_size = IR_CTX_HEADER_SIZE_CIP; + else + ctx_header_size = IR_CTX_HEADER_SIZE_NO_CIP; + maximum_usec_per_period = USEC_PER_SEC * PAGE_SIZE / + CYCLES_PER_SECOND / ctx_header_size; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 5000, UINT_MAX); + 5000, maximum_usec_per_period); if (err < 0) goto end; From f706df4f42b1b9f5acb6641eb6818d85b6856476 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 00:54:24 +0900 Subject: [PATCH 030/136] ALSA: firewire-lib: tune the minimum available size of PCM period In IEC 61883-1/6, one isoc packet can transfer events up to the value of syt interval. This comes from the interval of isoc cycle. As 1394 OHCI controller can generate hardware IRQ per isoc packet, the interval is calculated as 125 usec. In IEC 61883-1/6, two ways of transmission is described; blocking and non-blocking methods. In blocking method, the sequence of packet includes 'empty' or 'NODATA' packets which include no events. In non-blocking method, the number of events per packet is variable up to the syt interval. This commit uses double of the value of syt interval as minimum available size of PCM period due to the above protocol design. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191017155424.885-13-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 8a7818130382..7486eec4d958 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -213,9 +213,21 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, maximum_usec_per_period = USEC_PER_SEC * PAGE_SIZE / CYCLES_PER_SECOND / ctx_header_size; + // In IEC 61883-6, one isoc packet can transfer events up to the value + // of syt interval. This comes from the interval of isoc cycle. As 1394 + // OHCI controller can generate hardware IRQ per isoc packet, the + // interval is 125 usec. + // However, there are two ways of transmission in IEC 61883-6; blocking + // and non-blocking modes. In blocking mode, the sequence of isoc packet + // includes 'empty' or 'NODATA' packets which include no event. In + // non-blocking mode, the number of events per packet is variable up to + // the syt interval. + // Due to the above protocol design, the minimum PCM frames per + // interrupt should be double of the value of syt interval, thus it is + // 250 usec. err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 5000, maximum_usec_per_period); + 250, maximum_usec_per_period); if (err < 0) goto end; From 03b4816dcb6ee348edf248f7ba9f04f1e90c2fc7 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 15:19:06 +0900 Subject: [PATCH 031/136] ALSA: firewire-lib: add irq_target member into amdtp_domain struct This commit is a preparation to handle several IR/IT contexts in the same domain by tasklet context for one of the IT context. Such IT context is stored to AMDTP domain structure as 'irq_target'. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191018061911.24909-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 344818e936df..f92397a2f35f 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -279,6 +279,8 @@ struct amdtp_domain { unsigned int events_per_period; unsigned int events_per_buffer; + + struct amdtp_stream *irq_target; }; int amdtp_domain_init(struct amdtp_domain *d); From f890f9a04b361b2209c38e3317e6290a98e6ff6e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 15:19:07 +0900 Subject: [PATCH 032/136] ALSA: firewire-lib: replace pointer callback to flush isoc contexts in AMDTP domain An isoc context for AMDTP stream is flushed to queue packet by a call of pcm.pointer. This commit extends this for AMDTP domain. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191018061911.24909-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 51 ++++++++++++++---------- sound/firewire/amdtp-stream.h | 3 +- sound/firewire/bebob/bebob_pcm.c | 14 ++++--- sound/firewire/dice/dice-pcm.c | 4 +- sound/firewire/digi00x/digi00x-pcm.c | 4 +- sound/firewire/fireface/ff-pcm.c | 4 +- sound/firewire/fireworks/fireworks_pcm.c | 6 ++- sound/firewire/motu/motu-pcm.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 4 +- sound/firewire/tascam/tascam-pcm.c | 4 +- 10 files changed, 56 insertions(+), 42 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 7486eec4d958..23677b805b05 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1099,35 +1099,44 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, } /** - * amdtp_stream_pcm_pointer - get the PCM buffer position + * amdtp_domain_stream_pcm_pointer - get the PCM buffer position + * @d: the AMDTP domain. * @s: the AMDTP stream that transports the PCM data * * Returns the current buffer position, in frames. */ -unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) +unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, + struct amdtp_stream *s) { - /* - * This function is called in software IRQ context of period_tasklet or - * process context. - * - * When the software IRQ context was scheduled by software IRQ context - * of IR/IT contexts, queued packets were already handled. Therefore, - * no need to flush the queue in buffer anymore. - * - * When the process context reach here, some packets will be already - * queued in the buffer. These packets should be handled immediately - * to keep better granularity of PCM pointer. - * - * Later, the process context will sometimes schedules software IRQ - * context of the period_tasklet. Then, no need to flush the queue by - * the same reason as described for IR/IT contexts. - */ - if (!in_interrupt() && amdtp_stream_running(s)) - fw_iso_context_flush_completions(s->context); + struct amdtp_stream *irq_target = d->irq_target; + + if (irq_target && amdtp_stream_running(irq_target)) { + // This function is called in software IRQ context of + // period_tasklet or process context. + // + // When the software IRQ context was scheduled by software IRQ + // context of IT contexts, queued packets were already handled. + // Therefore, no need to flush the queue in buffer furthermore. + // + // When the process context reach here, some packets will be + // already queued in the buffer. These packets should be handled + // immediately to keep better granularity of PCM pointer. + // + // Later, the process context will sometimes schedules software + // IRQ context of the period_tasklet. Then, no need to flush the + // queue by the same reason as described in the above + if (!in_interrupt()) { + // Queued packet should be processed without any kernel + // preemption to keep latency against bus cycle. + preempt_disable(); + fw_iso_context_flush_completions(irq_target->context); + preempt_enable(); + } + } return READ_ONCE(s->pcm_buffer_pointer); } -EXPORT_SYMBOL(amdtp_stream_pcm_pointer); +EXPORT_SYMBOL_GPL(amdtp_domain_stream_pcm_pointer); /** * amdtp_stream_pcm_ack - acknowledge queued PCM frames diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index f92397a2f35f..ba0bbeddfdcb 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -198,7 +198,6 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); void amdtp_stream_pcm_prepare(struct amdtp_stream *s); -unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s); int amdtp_stream_pcm_ack(struct amdtp_stream *s); void amdtp_stream_pcm_abort(struct amdtp_stream *s); @@ -302,4 +301,6 @@ static inline int amdtp_domain_set_events_per_period(struct amdtp_domain *d, return 0; } +unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, + struct amdtp_stream *s); #endif diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 8b2e0ceffe82..dc15ea8d0dc5 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -313,17 +313,19 @@ pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static snd_pcm_uframes_t -pcm_capture_pointer(struct snd_pcm_substream *sbstrm) +static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) { struct snd_bebob *bebob = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&bebob->tx_stream); + + return amdtp_domain_stream_pcm_pointer(&bebob->domain, + &bebob->tx_stream); } -static snd_pcm_uframes_t -pcm_playback_pointer(struct snd_pcm_substream *sbstrm) +static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) { struct snd_bebob *bebob = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&bebob->rx_stream); + + return amdtp_domain_stream_pcm_pointer(&bebob->domain, + &bebob->rx_stream); } static int pcm_capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 7c0c34c5bd47..81b78e7d181d 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -379,14 +379,14 @@ static snd_pcm_uframes_t capture_pointer(struct snd_pcm_substream *substream) struct snd_dice *dice = substream->private_data; struct amdtp_stream *stream = &dice->tx_stream[substream->pcm->device]; - return amdtp_stream_pcm_pointer(stream); + return amdtp_domain_stream_pcm_pointer(&dice->domain, stream); } static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; struct amdtp_stream *stream = &dice->rx_stream[substream->pcm->device]; - return amdtp_stream_pcm_pointer(stream); + return amdtp_domain_stream_pcm_pointer(&dice->domain, stream); } static int capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index c9a833dff20d..f6a2053d5f10 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -301,14 +301,14 @@ static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) { struct snd_dg00x *dg00x = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&dg00x->tx_stream); + return amdtp_domain_stream_pcm_pointer(&dg00x->domain, &dg00x->tx_stream); } static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) { struct snd_dg00x *dg00x = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&dg00x->rx_stream); + return amdtp_domain_stream_pcm_pointer(&dg00x->domain, &dg00x->rx_stream); } static int pcm_capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index 005d959f8651..5af1dce90921 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -341,14 +341,14 @@ static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) { struct snd_ff *ff = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&ff->tx_stream); + return amdtp_domain_stream_pcm_pointer(&ff->domain, &ff->tx_stream); } static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) { struct snd_ff *ff = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&ff->rx_stream); + return amdtp_domain_stream_pcm_pointer(&ff->domain, &ff->rx_stream); } static int pcm_capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index abcc53dac8a5..71f5057caa0d 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -348,12 +348,14 @@ static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) { struct snd_efw *efw = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&efw->tx_stream); + + return amdtp_domain_stream_pcm_pointer(&efw->domain, &efw->tx_stream); } static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) { struct snd_efw *efw = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&efw->rx_stream); + + return amdtp_domain_stream_pcm_pointer(&efw->domain, &efw->rx_stream); } static int pcm_capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 00e693da0cad..13e2577c2a07 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -320,13 +320,13 @@ static snd_pcm_uframes_t capture_pointer(struct snd_pcm_substream *substream) { struct snd_motu *motu = substream->private_data; - return amdtp_stream_pcm_pointer(&motu->tx_stream); + return amdtp_domain_stream_pcm_pointer(&motu->domain, &motu->tx_stream); } static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream) { struct snd_motu *motu = substream->private_data; - return amdtp_stream_pcm_pointer(&motu->rx_stream); + return amdtp_domain_stream_pcm_pointer(&motu->domain, &motu->rx_stream); } static int capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index ba586d1ac91d..3be35dfcf270 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -393,13 +393,13 @@ static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstm) { struct snd_oxfw *oxfw = sbstm->private_data; - return amdtp_stream_pcm_pointer(&oxfw->tx_stream); + return amdtp_domain_stream_pcm_pointer(&oxfw->domain, &oxfw->tx_stream); } static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstm) { struct snd_oxfw *oxfw = sbstm->private_data; - return amdtp_stream_pcm_pointer(&oxfw->rx_stream); + return amdtp_domain_stream_pcm_pointer(&oxfw->domain, &oxfw->rx_stream); } static int pcm_capture_ack(struct snd_pcm_substream *substream) diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index b18664fdf955..1f66c8be7528 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -230,14 +230,14 @@ static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm) { struct snd_tscm *tscm = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&tscm->tx_stream); + return amdtp_domain_stream_pcm_pointer(&tscm->domain, &tscm->tx_stream); } static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm) { struct snd_tscm *tscm = sbstrm->private_data; - return amdtp_stream_pcm_pointer(&tscm->rx_stream); + return amdtp_domain_stream_pcm_pointer(&tscm->domain, &tscm->rx_stream); } static int pcm_capture_ack(struct snd_pcm_substream *substream) From e6dcc92fcea07f23c13996b7bb6a2a32ed4c6ed5 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 15:19:08 +0900 Subject: [PATCH 033/136] ALSA: firewire-lib: replace ack callback to flush isoc contexts in AMDTP domain An isoc context for AMDTP stream is flushed to queue packet by a call of pcm.ack. This commit extends this for AMDTP domain. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191018061911.24909-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 24 +++++++++++++++--------- sound/firewire/amdtp-stream.h | 3 ++- sound/firewire/bebob/bebob_pcm.c | 4 ++-- sound/firewire/dice/dice-pcm.c | 4 ++-- sound/firewire/digi00x/digi00x-pcm.c | 4 ++-- sound/firewire/fireface/ff-pcm.c | 4 ++-- sound/firewire/fireworks/fireworks_pcm.c | 4 ++-- sound/firewire/motu/motu-pcm.c | 4 ++-- sound/firewire/oxfw/oxfw-pcm.c | 4 ++-- sound/firewire/tascam/tascam-pcm.c | 4 ++-- 10 files changed, 33 insertions(+), 26 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 23677b805b05..3d27d4ce2b45 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1139,23 +1139,29 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, EXPORT_SYMBOL_GPL(amdtp_domain_stream_pcm_pointer); /** - * amdtp_stream_pcm_ack - acknowledge queued PCM frames + * amdtp_domain_stream_pcm_ack - acknowledge queued PCM frames + * @d: the AMDTP domain. * @s: the AMDTP stream that transfers the PCM frames * * Returns zero always. */ -int amdtp_stream_pcm_ack(struct amdtp_stream *s) +int amdtp_domain_stream_pcm_ack(struct amdtp_domain *d, struct amdtp_stream *s) { - /* - * Process isochronous packets for recent isochronous cycle to handle - * queued PCM frames. - */ - if (amdtp_stream_running(s)) - fw_iso_context_flush_completions(s->context); + struct amdtp_stream *irq_target = d->irq_target; + + // Process isochronous packets for recent isochronous cycle to handle + // queued PCM frames. + if (irq_target && amdtp_stream_running(irq_target)) { + // Queued packet should be processed without any kernel + // preemption to keep latency against bus cycle. + preempt_disable(); + fw_iso_context_flush_completions(irq_target->context); + preempt_enable(); + } return 0; } -EXPORT_SYMBOL(amdtp_stream_pcm_ack); +EXPORT_SYMBOL_GPL(amdtp_domain_stream_pcm_ack); /** * amdtp_stream_update - update the stream after a bus reset diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index ba0bbeddfdcb..470e77ca0061 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -198,7 +198,6 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); void amdtp_stream_pcm_prepare(struct amdtp_stream *s); -int amdtp_stream_pcm_ack(struct amdtp_stream *s); void amdtp_stream_pcm_abort(struct amdtp_stream *s); extern const unsigned int amdtp_syt_intervals[CIP_SFC_COUNT]; @@ -303,4 +302,6 @@ static inline int amdtp_domain_set_events_per_period(struct amdtp_domain *d, unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, struct amdtp_stream *s); +int amdtp_domain_stream_pcm_ack(struct amdtp_domain *d, struct amdtp_stream *s); + #endif diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index dc15ea8d0dc5..1b100159f4c5 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -332,14 +332,14 @@ static int pcm_capture_ack(struct snd_pcm_substream *substream) { struct snd_bebob *bebob = substream->private_data; - return amdtp_stream_pcm_ack(&bebob->tx_stream); + return amdtp_domain_stream_pcm_ack(&bebob->domain, &bebob->tx_stream); } static int pcm_playback_ack(struct snd_pcm_substream *substream) { struct snd_bebob *bebob = substream->private_data; - return amdtp_stream_pcm_ack(&bebob->rx_stream); + return amdtp_domain_stream_pcm_ack(&bebob->domain, &bebob->rx_stream); } int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 81b78e7d181d..f1848fb39bd0 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -394,7 +394,7 @@ static int capture_ack(struct snd_pcm_substream *substream) struct snd_dice *dice = substream->private_data; struct amdtp_stream *stream = &dice->tx_stream[substream->pcm->device]; - return amdtp_stream_pcm_ack(stream); + return amdtp_domain_stream_pcm_ack(&dice->domain, stream); } static int playback_ack(struct snd_pcm_substream *substream) @@ -402,7 +402,7 @@ static int playback_ack(struct snd_pcm_substream *substream) struct snd_dice *dice = substream->private_data; struct amdtp_stream *stream = &dice->rx_stream[substream->pcm->device]; - return amdtp_stream_pcm_ack(stream); + return amdtp_domain_stream_pcm_ack(&dice->domain, stream); } int snd_dice_create_pcm(struct snd_dice *dice) diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index f6a2053d5f10..8befc5d2ef22 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -315,14 +315,14 @@ static int pcm_capture_ack(struct snd_pcm_substream *substream) { struct snd_dg00x *dg00x = substream->private_data; - return amdtp_stream_pcm_ack(&dg00x->tx_stream); + return amdtp_domain_stream_pcm_ack(&dg00x->domain, &dg00x->tx_stream); } static int pcm_playback_ack(struct snd_pcm_substream *substream) { struct snd_dg00x *dg00x = substream->private_data; - return amdtp_stream_pcm_ack(&dg00x->rx_stream); + return amdtp_domain_stream_pcm_ack(&dg00x->domain, &dg00x->rx_stream); } int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index 5af1dce90921..c29f87a65c0f 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -355,14 +355,14 @@ static int pcm_capture_ack(struct snd_pcm_substream *substream) { struct snd_ff *ff = substream->private_data; - return amdtp_stream_pcm_ack(&ff->tx_stream); + return amdtp_domain_stream_pcm_ack(&ff->domain, &ff->tx_stream); } static int pcm_playback_ack(struct snd_pcm_substream *substream) { struct snd_ff *ff = substream->private_data; - return amdtp_stream_pcm_ack(&ff->rx_stream); + return amdtp_domain_stream_pcm_ack(&ff->domain, &ff->rx_stream); } int snd_ff_create_pcm_devices(struct snd_ff *ff) diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 71f5057caa0d..64c1bcf28dfa 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -362,14 +362,14 @@ static int pcm_capture_ack(struct snd_pcm_substream *substream) { struct snd_efw *efw = substream->private_data; - return amdtp_stream_pcm_ack(&efw->tx_stream); + return amdtp_domain_stream_pcm_ack(&efw->domain, &efw->tx_stream); } static int pcm_playback_ack(struct snd_pcm_substream *substream) { struct snd_efw *efw = substream->private_data; - return amdtp_stream_pcm_ack(&efw->rx_stream); + return amdtp_domain_stream_pcm_ack(&efw->domain, &efw->rx_stream); } int snd_efw_create_pcm_devices(struct snd_efw *efw) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 13e2577c2a07..55d3d6661731 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -333,14 +333,14 @@ static int capture_ack(struct snd_pcm_substream *substream) { struct snd_motu *motu = substream->private_data; - return amdtp_stream_pcm_ack(&motu->tx_stream); + return amdtp_domain_stream_pcm_ack(&motu->domain, &motu->tx_stream); } static int playback_ack(struct snd_pcm_substream *substream) { struct snd_motu *motu = substream->private_data; - return amdtp_stream_pcm_ack(&motu->rx_stream); + return amdtp_domain_stream_pcm_ack(&motu->domain, &motu->rx_stream); } int snd_motu_create_pcm_devices(struct snd_motu *motu) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 3be35dfcf270..74bd1811cec2 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -406,14 +406,14 @@ static int pcm_capture_ack(struct snd_pcm_substream *substream) { struct snd_oxfw *oxfw = substream->private_data; - return amdtp_stream_pcm_ack(&oxfw->tx_stream); + return amdtp_domain_stream_pcm_ack(&oxfw->domain, &oxfw->tx_stream); } static int pcm_playback_ack(struct snd_pcm_substream *substream) { struct snd_oxfw *oxfw = substream->private_data; - return amdtp_stream_pcm_ack(&oxfw->rx_stream); + return amdtp_domain_stream_pcm_ack(&oxfw->domain, &oxfw->rx_stream); } int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 1f66c8be7528..cd45f20ba515 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -244,14 +244,14 @@ static int pcm_capture_ack(struct snd_pcm_substream *substream) { struct snd_tscm *tscm = substream->private_data; - return amdtp_stream_pcm_ack(&tscm->tx_stream); + return amdtp_domain_stream_pcm_ack(&tscm->domain, &tscm->tx_stream); } static int pcm_playback_ack(struct snd_pcm_substream *substream) { struct snd_tscm *tscm = substream->private_data; - return amdtp_stream_pcm_ack(&tscm->rx_stream); + return amdtp_domain_stream_pcm_ack(&tscm->domain, &tscm->rx_stream); } int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) From 813dfbd6845e5366f2dbef190a695f8b06c1d83d Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 15:19:09 +0900 Subject: [PATCH 034/136] ALSA: firewire-lib: cancel flushing isoc context in the laste step to process context callback The aim of AMDTP domain is to process several isoc context in the same time. However, current implementation is against this idea because it flushes each isoc context in the end of processing context callback. This commit cancels it. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191018061911.24909-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 3d27d4ce2b45..36c3f1f9dbff 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -842,8 +842,6 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, } s->event_count = event_count; - - fw_iso_context_queue_flush(s->context); } static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, @@ -897,8 +895,6 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, } s->event_count = event_count; - - fw_iso_context_queue_flush(s->context); } /* this is executed one time */ From 60dd49298ec5803ca423836d5a724c3fe402cc3f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 15:19:10 +0900 Subject: [PATCH 035/136] ALSA: firewire-lib: handle several AMDTP streams in callback handler of IRQ target This commit changes AMDTP domain to run on an IT context of 1394 OHCI as IRQ target. No hardware interrupt is scheduled for the other isoc contexts. All of the isoc context are processed in a callback for an isoc context of IRQ target. The IRQ target is automatically selected from a list of AMDTP streams, thus users of AMDTP domain should add an AMDTP stream for IT context at least. The reason to select IT context as IRQ target is that the IT context runs on local 1394 OHCI controller and it can be used as reliable, constant IRQ generator. On the other hand, IR context can include skip cycle according to isoc packet transferred by device. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191018061911.24909-6-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 181 +++++++++++++++++++++++++--------- sound/firewire/amdtp-stream.h | 7 +- 2 files changed, 141 insertions(+), 47 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 36c3f1f9dbff..48be31eae9a5 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -482,13 +482,13 @@ static inline int queue_out_packet(struct amdtp_stream *s, } static inline int queue_in_packet(struct amdtp_stream *s, - struct fw_iso_packet *params, bool sched_irq) + struct fw_iso_packet *params) { // Queue one packet for IR context. params->header_length = s->ctx_data.tx.ctx_header_size; params->payload_length = s->ctx_data.tx.max_ctx_payload_length; params->skip = false; - return queue_packet(s, params, sched_irq); + return queue_packet(s, params, false); } static void generate_cip_header(struct amdtp_stream *s, __be32 cip_header[2], @@ -790,15 +790,24 @@ static void process_ctx_payloads(struct amdtp_stream *s, update_pcm_pointers(s, pcm, pcm_frames); } +static void amdtp_stream_master_callback(struct fw_iso_context *context, + u32 tstamp, size_t header_length, + void *header, void *private_data); + +static void amdtp_stream_master_first_callback(struct fw_iso_context *context, + u32 tstamp, size_t header_length, + void *header, void *private_data); + static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; const __be32 *ctx_header = header; - unsigned int events_per_period = s->events_per_period; - unsigned int event_count = s->event_count; + unsigned int events_per_period = s->ctx_data.rx.events_per_period; + unsigned int event_count = s->ctx_data.rx.event_count; unsigned int packets; + bool is_irq_target; int i; if (s->packet_index < 0) @@ -811,6 +820,10 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, process_ctx_payloads(s, s->pkt_descs, packets); + is_irq_target = + !!(context->callback.sc == amdtp_stream_master_callback || + context->callback.sc == amdtp_stream_master_first_callback); + for (i = 0; i < packets; ++i) { const struct pkt_desc *desc = s->pkt_descs + i; unsigned int syt; @@ -829,10 +842,12 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, desc->data_blocks, desc->data_block_counter, syt, i); - event_count += desc->data_blocks; - if (event_count >= events_per_period) { - event_count -= events_per_period; - sched_irq = true; + if (is_irq_target) { + event_count += desc->data_blocks; + if (event_count >= events_per_period) { + event_count -= events_per_period; + sched_irq = true; + } } if (queue_out_packet(s, &template.params, sched_irq) < 0) { @@ -841,7 +856,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, } } - s->event_count = event_count; + s->ctx_data.rx.event_count = event_count; } static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, @@ -850,8 +865,6 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, { struct amdtp_stream *s = private_data; __be32 *ctx_header = header; - unsigned int events_per_period = s->events_per_period; - unsigned int event_count = s->event_count; unsigned int packets; int i; int err; @@ -873,31 +886,47 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, } for (i = 0; i < packets; ++i) { - const struct pkt_desc *desc = s->pkt_descs + i; struct fw_iso_packet params = {0}; - bool sched_irq = false; - if (err >= 0) { - event_count += desc->data_blocks; - if (event_count >= events_per_period) { - event_count -= events_per_period; - sched_irq = true; - } - } else { - sched_irq = - !((s->packet_index + 1) % s->idle_irq_interval); - } - - if (queue_in_packet(s, ¶ms, sched_irq) < 0) { + if (queue_in_packet(s, ¶ms) < 0) { cancel_stream(s); return; } } - - s->event_count = event_count; } -/* this is executed one time */ +static void amdtp_stream_master_callback(struct fw_iso_context *context, + u32 tstamp, size_t header_length, + void *header, void *private_data) +{ + struct amdtp_domain *d = private_data; + struct amdtp_stream *irq_target = d->irq_target; + struct amdtp_stream *s; + + out_stream_callback(context, tstamp, header_length, header, irq_target); + if (amdtp_streaming_error(irq_target)) + goto error; + + list_for_each_entry(s, &d->streams, list) { + if (s != irq_target && amdtp_stream_running(s)) { + fw_iso_context_flush_completions(s->context); + if (amdtp_streaming_error(s)) + goto error; + } + } + + return; +error: + if (amdtp_stream_running(irq_target)) + cancel_stream(irq_target); + + list_for_each_entry(s, &d->streams, list) { + if (amdtp_stream_running(s)) + cancel_stream(s); + } +} + +// this is executed one time. static void amdtp_stream_first_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) @@ -928,18 +957,39 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, context->callback.sc(context, tstamp, header_length, header, s); } +static void amdtp_stream_master_first_callback(struct fw_iso_context *context, + u32 tstamp, size_t header_length, + void *header, void *private_data) +{ + struct amdtp_domain *d = private_data; + struct amdtp_stream *s = d->irq_target; + const __be32 *ctx_header = header; + + s->callbacked = true; + wake_up(&s->callback_wait); + + s->start_cycle = compute_it_cycle(*ctx_header, s->queue_size); + + context->callback.sc = amdtp_stream_master_callback; + + context->callback.sc(context, tstamp, header_length, header, d); +} + /** * amdtp_stream_start - start transferring packets * @s: the AMDTP stream to start * @channel: the isochronous channel on the bus * @speed: firewire speed code + * @d: the AMDTP domain to which the AMDTP stream belongs + * @is_irq_target: whether isoc context for the AMDTP stream is used to generate + * hardware IRQ. * * The stream cannot be started until it has been configured with * amdtp_stream_set_parameters() and it must be started before any PCM or MIDI * device can be started. */ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, - struct amdtp_domain *d) + struct amdtp_domain *d, bool is_irq_target) { static const struct { unsigned int data_block; @@ -955,10 +1005,13 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, }; unsigned int events_per_buffer = d->events_per_buffer; unsigned int events_per_period = d->events_per_period; + unsigned int idle_irq_interval; unsigned int ctx_header_size; unsigned int max_ctx_payload_size; enum dma_data_direction dir; int type, tag, err; + fw_iso_callback_t ctx_cb; + void *ctx_data; mutex_lock(&s->mutex); @@ -969,6 +1022,12 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, } if (s->direction == AMDTP_IN_STREAM) { + // NOTE: IT context should be used for constant IRQ. + if (is_irq_target) { + err = -EINVAL; + goto err_unlock; + } + s->data_block_counter = UINT_MAX; } else { entry = &initial_state[s->sfc]; @@ -1008,22 +1067,29 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, if (events_per_buffer == 0) events_per_buffer = events_per_period * 3; - s->idle_irq_interval = - DIV_ROUND_UP(CYCLES_PER_SECOND * events_per_period, - amdtp_rate_table[s->sfc]); + idle_irq_interval = DIV_ROUND_UP(CYCLES_PER_SECOND * events_per_period, + amdtp_rate_table[s->sfc]); s->queue_size = DIV_ROUND_UP(CYCLES_PER_SECOND * events_per_buffer, amdtp_rate_table[s->sfc]); - s->events_per_period = events_per_period; - s->event_count = 0; err = iso_packets_buffer_init(&s->buffer, s->unit, s->queue_size, max_ctx_payload_size, dir); if (err < 0) goto err_unlock; + if (is_irq_target) { + s->ctx_data.rx.events_per_period = events_per_period; + s->ctx_data.rx.event_count = 0; + ctx_cb = amdtp_stream_master_first_callback; + ctx_data = d; + } else { + ctx_cb = amdtp_stream_first_callback; + ctx_data = s; + } + s->context = fw_iso_context_create(fw_parent_device(s->unit)->card, type, channel, speed, ctx_header_size, - amdtp_stream_first_callback, s); + ctx_cb, ctx_data); if (IS_ERR(s->context)) { err = PTR_ERR(s->context); if (err == -EBUSY) @@ -1054,14 +1120,20 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, s->packet_index = 0; do { struct fw_iso_packet params; - bool sched_irq; - sched_irq = !((s->packet_index + 1) % s->idle_irq_interval); if (s->direction == AMDTP_IN_STREAM) { - err = queue_in_packet(s, ¶ms, sched_irq); + err = queue_in_packet(s, ¶ms); } else { + bool sched_irq = false; + params.header_length = 0; params.payload_length = 0; + + if (is_irq_target) { + sched_irq = !((s->packet_index + 1) % + idle_irq_interval); + } + err = queue_out_packet(s, ¶ms, sched_irq); } if (err < 0) @@ -1276,17 +1348,33 @@ int amdtp_domain_start(struct amdtp_domain *d) struct amdtp_stream *s; int err = 0; + // Select an IT context as IRQ target. list_for_each_entry(s, &d->streams, list) { - err = amdtp_stream_start(s, s->channel, s->speed, d); - if (err < 0) + if (s->direction == AMDTP_OUT_STREAM) break; } + if (!s) + return -ENXIO; + d->irq_target = s; - if (err < 0) { - list_for_each_entry(s, &d->streams, list) - amdtp_stream_stop(s); + list_for_each_entry(s, &d->streams, list) { + if (s != d->irq_target) { + err = amdtp_stream_start(s, s->channel, s->speed, d, + false); + if (err < 0) + goto error; + } } + s = d->irq_target; + err = amdtp_stream_start(s, s->channel, s->speed, d, true); + if (err < 0) + goto error; + + return 0; +error: + list_for_each_entry(s, &d->streams, list) + amdtp_stream_stop(s); return err; } EXPORT_SYMBOL_GPL(amdtp_domain_start); @@ -1299,12 +1387,17 @@ void amdtp_domain_stop(struct amdtp_domain *d) { struct amdtp_stream *s, *next; + if (d->irq_target) + amdtp_stream_stop(d->irq_target); + list_for_each_entry_safe(s, next, &d->streams, list) { list_del(&s->list); - amdtp_stream_stop(s); + if (s != d->irq_target) + amdtp_stream_stop(s); } d->events_per_period = 0; + d->irq_target = NULL; } EXPORT_SYMBOL_GPL(amdtp_domain_stop); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 470e77ca0061..c4bde69c2d4f 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -143,11 +143,12 @@ struct amdtp_stream { // To generate CIP header. unsigned int fdf; int syt_override; + + // To generate constant hardware IRQ. + unsigned int event_count; + unsigned int events_per_period; } rx; } ctx_data; - unsigned int event_count; - unsigned int events_per_period; - unsigned int idle_irq_interval; /* For CIP headers. */ unsigned int source_node_id_field; From acfedcbe1ce4c69e1da914f39c02d945c80198d4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 18 Oct 2019 15:19:11 +0900 Subject: [PATCH 036/136] ALSA: firewire-lib: postpone to start IR context Some devices have a quirk to postpone transmission of isoc packet for several dozen or hundred isoc cycles since configured to transmit. Furthermore, some devices have a quirk to transmit isoc packet with discontinued data of its header. In 1394 OHCI specification, software allows to start isoc context with certain isoc cycle. Linux firewire subsystem has kernel API to use it as well. This commit uses the functionality of 1394 OHCI controller to handle the quirks. At present, this feature is convenient to ALSA bebob and fireface driver. As a result, some devices can be safely handled, as long as I know: - MAudio FireWire solo - MAudio ProFire Lightbridge - MAudio FireWire 410 - Roland FA-66 Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191018061911.24909-7-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 79 +++++++++++++++++++-- sound/firewire/amdtp-stream.h | 2 +- sound/firewire/bebob/bebob_stream.c | 10 ++- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/digi00x/digi00x-stream.c | 2 +- sound/firewire/fireface/ff-stream.c | 10 ++- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/motu-stream.c | 2 +- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/tascam/tascam-stream.c | 2 +- 10 files changed, 98 insertions(+), 15 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 48be31eae9a5..37d38efb4c87 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -9,6 +9,7 @@ #include #include #include +#include #include #include #include @@ -983,13 +984,16 @@ static void amdtp_stream_master_first_callback(struct fw_iso_context *context, * @d: the AMDTP domain to which the AMDTP stream belongs * @is_irq_target: whether isoc context for the AMDTP stream is used to generate * hardware IRQ. + * @start_cycle: the isochronous cycle to start the context. Start immediately + * if negative value is given. * * The stream cannot be started until it has been configured with * amdtp_stream_set_parameters() and it must be started before any PCM or MIDI * device can be started. */ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, - struct amdtp_domain *d, bool is_irq_target) + struct amdtp_domain *d, bool is_irq_target, + int start_cycle) { static const struct { unsigned int data_block; @@ -1146,7 +1150,7 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed, tag |= FW_ISO_CONTEXT_MATCH_TAG0; s->callbacked = false; - err = fw_iso_context_start(s->context, -1, 0, tag); + err = fw_iso_context_start(s->context, start_cycle, 0, tag); if (err < 0) goto err_pkt_descs; @@ -1339,14 +1343,40 @@ int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s, } EXPORT_SYMBOL_GPL(amdtp_domain_add_stream); +static int get_current_cycle_time(struct fw_card *fw_card, int *cur_cycle) +{ + int generation; + int rcode; + __be32 reg; + u32 data; + + // This is a request to local 1394 OHCI controller and expected to + // complete without any event waiting. + generation = fw_card->generation; + smp_rmb(); // node_id vs. generation. + rcode = fw_run_transaction(fw_card, TCODE_READ_QUADLET_REQUEST, + fw_card->node_id, generation, SCODE_100, + CSR_REGISTER_BASE + CSR_CYCLE_TIME, + ®, sizeof(reg)); + if (rcode != RCODE_COMPLETE) + return -EIO; + + data = be32_to_cpu(reg); + *cur_cycle = data >> 12; + + return 0; +} + /** * amdtp_domain_start - start sending packets for isoc context in the domain. * @d: the AMDTP domain. + * @ir_delay_cycle: the cycle delay to start all IR contexts. */ -int amdtp_domain_start(struct amdtp_domain *d) +int amdtp_domain_start(struct amdtp_domain *d, unsigned int ir_delay_cycle) { struct amdtp_stream *s; - int err = 0; + int cycle; + int err; // Select an IT context as IRQ target. list_for_each_entry(s, &d->streams, list) { @@ -1357,17 +1387,54 @@ int amdtp_domain_start(struct amdtp_domain *d) return -ENXIO; d->irq_target = s; + if (ir_delay_cycle > 0) { + struct fw_card *fw_card = fw_parent_device(s->unit)->card; + + err = get_current_cycle_time(fw_card, &cycle); + if (err < 0) + return err; + + // No need to care overflow in cycle field because of enough + // width. + cycle += ir_delay_cycle; + + // Round up to sec field. + if ((cycle & 0x00001fff) >= CYCLES_PER_SECOND) { + unsigned int sec; + + // The sec field can overflow. + sec = (cycle & 0xffffe000) >> 13; + cycle = (++sec << 13) | + ((cycle & 0x00001fff) / CYCLES_PER_SECOND); + } + + // In OHCI 1394 specification, lower 2 bits are available for + // sec field. + cycle &= 0x00007fff; + } else { + cycle = -1; + } + list_for_each_entry(s, &d->streams, list) { + int cycle_match; + + if (s->direction == AMDTP_IN_STREAM) { + cycle_match = cycle; + } else { + // IT context starts immediately. + cycle_match = -1; + } + if (s != d->irq_target) { err = amdtp_stream_start(s, s->channel, s->speed, d, - false); + false, cycle_match); if (err < 0) goto error; } } s = d->irq_target; - err = amdtp_stream_start(s, s->channel, s->speed, d, true); + err = amdtp_stream_start(s, s->channel, s->speed, d, true, -1); if (err < 0) goto error; diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index c4bde69c2d4f..f2d44e2dc3c8 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -288,7 +288,7 @@ void amdtp_domain_destroy(struct amdtp_domain *d); int amdtp_domain_add_stream(struct amdtp_domain *d, struct amdtp_stream *s, int channel, int speed); -int amdtp_domain_start(struct amdtp_domain *d); +int amdtp_domain_start(struct amdtp_domain *d, unsigned int ir_delay_cycle); void amdtp_domain_stop(struct amdtp_domain *d); static inline int amdtp_domain_set_events_per_period(struct amdtp_domain *d, diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 5e4a61458be2..7ac0d9f495c4 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -658,7 +658,15 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) if (err < 0) goto error; - err = amdtp_domain_start(&bebob->domain); + // The device postpones start of transmission mostly for 1 sec + // after receives packets firstly. For safe, IR context starts + // 1.5 sec (=12000 cycles) later. This is within 2.0 sec + // (=CALLBACK_TIMEOUT). + // Furthermore, some devices transfer isoc packets with + // discontinuous counter in the beginning of packet streaming. + // The delay has an effect to avoid detection of this + // discontinuity. + err = amdtp_domain_start(&bebob->domain, 12000); if (err < 0) goto error; diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 0cff346e8052..6a3d60913e10 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -462,7 +462,7 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice) goto error; } - err = amdtp_domain_start(&dice->domain); + err = amdtp_domain_start(&dice->domain, 0); if (err < 0) goto error; diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index 0c539188ba18..405d6903bfbc 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -375,7 +375,7 @@ int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x) if (err < 0) goto error; - err = amdtp_domain_start(&dg00x->domain); + err = amdtp_domain_start(&dg00x->domain, 0); if (err < 0) goto error; diff --git a/sound/firewire/fireface/ff-stream.c b/sound/firewire/fireface/ff-stream.c index a13754f914e8..63b79c4a5405 100644 --- a/sound/firewire/fireface/ff-stream.c +++ b/sound/firewire/fireface/ff-stream.c @@ -184,6 +184,7 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) */ if (!amdtp_stream_running(&ff->rx_stream)) { int spd = fw_parent_device(ff->unit)->max_speed; + unsigned int ir_delay_cycle; err = ff->spec->protocol->begin_session(ff, rate); if (err < 0) @@ -199,7 +200,14 @@ int snd_ff_stream_start_duplex(struct snd_ff *ff, unsigned int rate) if (err < 0) goto error; - err = amdtp_domain_start(&ff->domain); + // The device postpones start of transmission mostly for several + // cycles after receiving packets firstly. + if (ff->spec->protocol == &snd_ff_protocol_ff800) + ir_delay_cycle = 800; // = 100 msec + else + ir_delay_cycle = 16; // = 2 msec + + err = amdtp_domain_start(&ff->domain, ir_delay_cycle); if (err < 0) goto error; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index f35a33d4d4e6..2206af0fef42 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -272,7 +272,7 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw) if (err < 0) goto error; - err = amdtp_domain_start(&efw->domain); + err = amdtp_domain_start(&efw->domain, 0); if (err < 0) goto error; diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 9975770c9b1f..a17ddceb1bec 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -260,7 +260,7 @@ int snd_motu_stream_start_duplex(struct snd_motu *motu) if (err < 0) goto stop_streams; - err = amdtp_domain_start(&motu->domain); + err = amdtp_domain_start(&motu->domain, 0); if (err < 0) goto stop_streams; diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 995e9c5bd84b..501a80094bf7 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -355,7 +355,7 @@ int snd_oxfw_stream_start_duplex(struct snd_oxfw *oxfw) } } - err = amdtp_domain_start(&oxfw->domain); + err = amdtp_domain_start(&oxfw->domain, 0); if (err < 0) goto error; diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index a9b3b7eb6d21..eb07e1decf9b 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -473,7 +473,7 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) goto error; - err = amdtp_domain_start(&tscm->domain); + err = amdtp_domain_start(&tscm->domain, 0); if (err < 0) return err; From 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Fri, 18 Oct 2019 15:38:47 +0800 Subject: [PATCH 037/136] PCI: Add a helper to check Power Resource Requirements _PR3 existence A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Signed-off-by: Kai-Heng Feng Acked-by: Bjorn Helgaas Link: https://lore.kernel.org/r/20191018073848.14590-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- drivers/pci/pci.c | 18 ++++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 20 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index e7982af9a5d8..1df99d9e350e 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,24 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +#ifdef CONFIG_ACPI +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct acpi_device *adev; + + if (acpi_disabled) + return false; + + adev = ACPI_COMPANION(&pdev->dev); + if (!adev) + return false; + + return adev->power.flags.power_resources && + acpi_has_method(adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); +#endif + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index f9088c89a534..1d15c5d49cdd 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2310,9 +2310,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH From bacd861452d2be86a4df341b12e32db7dac8021e Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Fri, 18 Oct 2019 15:38:48 +0800 Subject: [PATCH 038/136] ALSA: hda: Allow HDA to be runtime suspended when dGPU is not bound to a driver Nvidia proprietary driver doesn't support runtime power management, so when a user only wants to use the integrated GPU, it's a common practice to let dGPU not to bind any driver, and let its upstream port to be runtime suspended. At the end of runtime suspension the port uses platform power management to disable power through _OFF method of power resource, which is listed by _PR3. After commit b516ea586d71 ("PCI: Enable NVIDIA HDA controllers"), when the dGPU comes with an HDA function, the HDA won't be suspended if the dGPU is unbound, so the power resource can't be turned off by its upstream port driver. Commit 37a3a98ef601 ("ALSA: hda - Enable runtime PM only for discrete GPU") only allows HDA to be runtime suspended once GPU is bound, to keep APU's HDA working. However, HDA on dGPU isn't that useful if dGPU is not bound to any driver. So let's relax the runtime suspend requirement for dGPU's HDA function, to disable the power source to save lots of power. BugLink: https://bugs.launchpad.net/bugs/1840835 Fixes: b516ea586d71 ("PCI: Enable NVIDIA HDA controllers") Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20191018073848.14590-2-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..e63b871343e5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1280,11 +1280,17 @@ static void init_vga_switcheroo(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct pci_dev *p = get_bound_vga(chip->pci); + struct pci_dev *parent; if (p) { dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; - chip->bus.keep_power = 1; /* cleared in either gpu_bound op or codec probe */ + + /* cleared in either gpu_bound op or codec probe, or when its + * upstream port has _PR3 (i.e. dGPU). + */ + parent = pci_upstream_bridge(p); + chip->bus.keep_power = parent ? !pci_pr3_present(parent) : 1; chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } From 46b4bff6572b0552b1ee062043621e4b252638d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Oct 2019 16:25:20 +0200 Subject: [PATCH 039/136] PCI: Fix missing inline for pci_pr3_present() The inline prefix was missing in the dummy function pci_pr3_present() definition. Fix it. Reported-by: kbuild test robot Fixes: 52525b7a3cf8 ("PCI: Add a helper to check Power Resource Requirements _PR3 existence") Link: https://lore.kernel.org/r/201910212111.qHm6OcWx%lkp@intel.com Signed-off-by: Takashi Iwai --- include/linux/pci.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/pci.h b/include/linux/pci.h index 1d15c5d49cdd..be529d311122 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2314,7 +2314,7 @@ bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } -static bool pci_pr3_present(struct pci_dev *pdev) { return false; } +static inline bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH From 82d9d54a6c0ee8b12211fa4e59fd940a2da4e063 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 22 Oct 2019 19:43:12 +0200 Subject: [PATCH 040/136] ALSA: hda: add Intel DSP configuration / probe code For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191022174313.29087-1-perex@perex.cz Signed-off-by: Takashi Iwai --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 5 +- sound/hda/intel-dsp-config.c | 249 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 49 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 322 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..601e617918b8 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,6 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +snd-intel-dspcfg-$(CONFIG_SND_INTEL_NHLT) += intel-nhlt.o +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..d9f6d9e872b4 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela + +#include +#include +#include +#include +#include +#include +#include + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table + * - the order of similar PCI ID entries is important! + * - the first successful match will win + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e63b871343e5..ca462dd39a48 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include #include #include -#include +#include #include #include #include @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool dsp_driver = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,9 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(dsp_driver, bool, 0444); +MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1)"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2026,25 +2027,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2062,6 +2044,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2075,17 +2067,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 597516c0c192..3f9e65c10d5b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include #include #include +#include #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 18c41fdc47c8..806dfa0e5eae 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ From cc8f81c7e625168a60843b2b39e3a327cf5170fe Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 22 Oct 2019 19:43:13 +0200 Subject: [PATCH 041/136] ALSA: hda: fix intel DSP config Reshuffle list of devices by historical order and add correct information as needed. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20191022174313.29087-2-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 276 ++++++++++++++++++++++++----------- 1 file changed, 192 insertions(+), 84 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index d9f6d9e872b4..0b2a7201a46a 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -30,31 +30,6 @@ struct config_entry { * - the first successful match will win */ static const struct config_entry config_table[] = { -/* Cometlake-LP */ -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) - { - /* prefer SST */ - .flags = FLAG_SST, - .device = 0x02c8, - }, -#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) - { - .flags = FLAG_SOF, - .device = 0x02c8, - }, -#endif -/* Cometlake-H */ -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) - { - .flags = FLAG_SST, - .device = 0x06c8, - }, -#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) - { - .flags = FLAG_SOF, - .device = 0x06c8, - }, -#endif /* Merrifield */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) { @@ -65,47 +40,15 @@ static const struct config_entry config_table[] = { /* Broxton-T */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF, .device = 0x1a98, }, #endif -/* Geminilake */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) - { - .flags = FLAG_SOF, - .device = 0x3198, - .dmi_table = (const struct dmi_system_id []) { - { - .ident = "Google Chromebooks", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Google"), - } - }, - {} - } - }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) - { - .flags = FLAG_SST, - .device = 0x3198, - }, -#endif -/* Icelake */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, - .device = 0x34c8, - }, -#endif -/* Elkhart Lake */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, - .device = 0x4b55, - }, -#endif -/* Appololake (Broxton-P) */ +/* + * Apollolake (Broxton-P) + * the legacy HDaudio driver is used except on Up Squared (SOF) and + * Chromebooks (SST) + */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) { .flags = FLAG_SOF, @@ -126,43 +69,208 @@ static const struct config_entry config_table[] = { { .flags = FLAG_SST, .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } }, #endif +/* + * Skylake and Kabylake use legacy HDaudio driver except for Google + * Chromebooks (SST) + */ + +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKL) + { + .flags = FLAG_SST, + .device = 0x9d70, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_KBL) + { + .flags = FLAG_SST, + .device = 0x9d71, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif + +/* + * Geminilake uses legacy HDaudio driver except for Google + * Chromebooks + */ +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif + +/* + * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy + * HDaudio driver except for Google Chromebooks and when DMICs are + * present. Two cases are required since Coreboot does not expose NHLT + * tables. + * + * When the Chromebook quirk is not present, it's based on information + * that no such device exists. When the quirk is present, it could be + * either based on product information or a placeholder. + */ + /* Cannonlake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF, + .device = 0x9dc8, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, .device = 0x9dc8, }, #endif -/* Sunrise Point-LP */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, - .device = 0x9d70, - }, -#endif -/* Kabylake-LP */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, - .device = 0x9d71, - }, -#endif -/* Tigerlake */ -#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, - .device = 0xa0c8, - }, -#endif + /* Coffelake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF, + .device = 0xa348, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, .device = 0xa348, }, #endif + +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x06c8, + }, +#endif + +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF, + .device = 0x34c8, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif + +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF, + .device = 0xa0c8, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif + +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif + }; static const struct config_entry *snd_intel_dsp_find_config From 0a671dc500553cf6f4cc3efbcb0923b5e9adccb5 Mon Sep 17 00:00:00 2001 From: YueHaibing Date: Fri, 25 Oct 2019 17:39:05 +0800 Subject: [PATCH 042/136] ALSA: intel_hdmi: Remove dev_err() on platform_get_irq() failure platform_get_irq() will call dev_err() itself on failure, so there is no need for the driver to also do this. This is detected by coccinelle. Signed-off-by: YueHaibing Link: https://lore.kernel.org/r/20191025093905.14888-1-yuehaibing@huawei.com Signed-off-by: Takashi Iwai --- sound/x86/intel_hdmi_audio.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 5fd4e32247a6..cd389d21219a 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1708,10 +1708,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* get resources */ irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "Could not get irq resource: %d\n", irq); + if (irq < 0) return irq; - } res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res_mmio) { From ca58f55108fee41d87c9123f85ad4863e5de7f45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2019 11:58:03 +0100 Subject: [PATCH 043/136] ALSA: hda - Fix pending unsol events at shutdown This is an alternative fix attemp for the issue reported in the commit caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was reverted later due to regressions. Instead of tweaking the hardware disablement order and the enforced irq flushing, do calling cancel_work_sync() of the unsol work early enough, and explicitly ignore the unsol events during the shutdown by checking the bus->shutdown flag. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") Cc: Chris Wilson Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 4 ++++ sound/pci/hda/hda_intel.c | 3 +++ 2 files changed, 7 insertions(+) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 8272b50b8349..6a8564566375 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -43,6 +43,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2a9d87ff2e1c..121b20121dae 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1389,8 +1389,11 @@ static int azx_free(struct azx *chip) static int azx_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; + struct hdac_bus *bus = azx_bus(chip); chip->bus.shutdown = 1; + cancel_work_sync(&bus->unsol_work); + return 0; } From 91636a82044a2821201b54faac4d1d2425260842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2019 14:06:34 +0100 Subject: [PATCH 044/136] ALSA: hda: Allow non-Intel device probe gracefully The recent addition of snd_intel_dsp_driver_probe() check caused a spurious kernel warning when the driver is loaded for a non-Intel hardware due to snd_BUG_ON(). Moreover, for such a hardware, we should always return SND_INTEL_DSP_DRIVER_ANY, not check the dsp_driver option at all. This patch fixes these issues for non-Intel devices. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20191028130634.3501-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 0b2a7201a46a..be1df80ed013 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,13 +307,13 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) { const struct config_entry *cfg; + /* Intel vendor only */ + if (pci->vendor != 0x8086) + return SND_INTEL_DSP_DRIVER_ANY; + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) return dsp_driver; - /* Intel vendor only */ - if (snd_BUG_ON(pci->vendor != 0x8086)) - return SND_INTEL_DSP_DRIVER_ANY; - /* * detect DSP by checking class/subclass/prog-id information * class=04 subclass 03 prog-if 00: no DSP, use legacy driver From 1d4961d9eb1aaa498dfb44779b7e4b95d79112d0 Mon Sep 17 00:00:00 2001 From: Saurav Girepunje Date: Tue, 29 Oct 2019 23:22:00 +0530 Subject: [PATCH 045/136] ALSA: usb-audio: sound: usb: usb true/false for bool return type Use true/false for bool type return in uac_clock_source_is_valid(). Signed-off-by: Saurav Girepunje Link: https://lore.kernel.org/r/20191029175200.GA7320@saurav Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 6b8c14f9b5d4..018b1ecb5404 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -165,21 +165,21 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = le32_to_cpu(cs_desc->bmControls); } else { /* UAC_VERSION_1/2 */ struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = cs_desc->bmControls; } /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) - return 1; + return true; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -191,10 +191,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, dev_warn(&dev->dev, "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return 0; + return false; } - return !!data; + return data ? true : false; } static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, From 2644df63c709e107c1882be95d802adea284dbf5 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Oct 2019 17:06:39 +0900 Subject: [PATCH 046/136] ALSA: firewire-motu: fix wrong spelling for macro Just replace 'SEPARETED' with 'SEPARATED' for macro. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030080644.1704-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 2 +- sound/firewire/motu/motu-protocol-v3.c | 2 +- sound/firewire/motu/motu.c | 8 ++++---- sound/firewire/motu/motu.h | 2 +- 4 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 9e2f16eebe0a..a42f77fc3858 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -191,7 +191,7 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } } else { - if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) { + if (flags & SND_MOTU_SPEC_RX_SEPARATED_MAIN) { pcm_chunks[0] += 2; pcm_chunks[1] += 2; } diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 5eafa506e8a9..ba5794c724be 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -187,7 +187,7 @@ static void calculate_fixed_part(struct snd_motu_packet_format *formats, pcm_chunks[1] += 2; } } else { - if (flags & SND_MOTU_SPEC_RX_SEPARETED_MAIN) { + if (flags & SND_MOTU_SPEC_RX_SEPARATED_MAIN) { pcm_chunks[0] += 2; pcm_chunks[1] += 2; } diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 72908b4de77c..680e363d0a4b 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -178,7 +178,7 @@ static const struct snd_motu_spec motu_828mk2 = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | - SND_MOTU_SPEC_RX_SEPARETED_MAIN | + SND_MOTU_SPEC_RX_SEPARATED_MAIN | SND_MOTU_SPEC_HAS_OPT_IFACE_A | SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_2ND_Q, @@ -224,7 +224,7 @@ static const struct snd_motu_spec motu_828mk3 = { SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | SND_MOTU_SPEC_TX_REVERB_CHUNK | - SND_MOTU_SPEC_RX_SEPARETED_MAIN | + SND_MOTU_SPEC_RX_SEPARATED_MAIN | SND_MOTU_SPEC_HAS_OPT_IFACE_A | SND_MOTU_SPEC_HAS_OPT_IFACE_B | SND_MOTU_SPEC_RX_MIDI_3RD_Q | @@ -240,7 +240,7 @@ static const struct snd_motu_spec motu_audio_express = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | - SND_MOTU_SPEC_RX_SEPARETED_MAIN | + SND_MOTU_SPEC_RX_SEPARATED_MAIN | SND_MOTU_SPEC_RX_MIDI_2ND_Q | SND_MOTU_SPEC_TX_MIDI_3RD_Q, .analog_in_ports = 2, @@ -253,7 +253,7 @@ static const struct snd_motu_spec motu_4pre = { .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | SND_MOTU_SPEC_TX_MICINST_CHUNK | SND_MOTU_SPEC_TX_RETURN_CHUNK | - SND_MOTU_SPEC_RX_SEPARETED_MAIN, + SND_MOTU_SPEC_RX_SEPARATED_MAIN, .analog_in_ports = 2, .analog_out_ports = 2, }; diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 35ed8b6bb8e8..08e3d568221d 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -86,7 +86,7 @@ enum snd_motu_spec_flags { SND_MOTU_SPEC_RX_MIDI_3RD_Q = 0x0200, SND_MOTU_SPEC_TX_MIDI_2ND_Q = 0x0400, SND_MOTU_SPEC_TX_MIDI_3RD_Q = 0x0800, - SND_MOTU_SPEC_RX_SEPARETED_MAIN = 0x1000, + SND_MOTU_SPEC_RX_SEPARATED_MAIN = 0x1000, }; #define SND_MOTU_CLOCK_RATE_COUNT 6 From 1ef2ff941424bde4fd1d8dc97e0e600c5dc5472e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Oct 2019 17:06:40 +0900 Subject: [PATCH 047/136] ALSA: firewire-motu: print for unknown source of sampling clock When unknown source is detected for sampling clock, corresponding label was not added for node on procfs. This commit adds it. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030080644.1704-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-proc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index ea46fb4c1b5a..4e5767c0c8d7 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -19,6 +19,7 @@ static const char *const clock_names[] = { [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface", [SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface", [SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface", + [SND_MOTU_CLOCK_SOURCE_UNKNOWN] = "Unknown", }; static void proc_read_clock(struct snd_info_entry *entry, From 3f58f004bff3273d102583e0b2f478526cf9d3c8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Oct 2019 17:06:41 +0900 Subject: [PATCH 048/136] ALSA: firewire-motu: detect SPH source of sampling clock In MOTU FireWire series, devices have a mode to generate sampling clock from a sequence of source packet header (SPH) included in each data block of received packet. This mode is used for several purposes such as mode for SMPTE time code, sync to the other sound cards and so on. This commit adds support for the SPH mode. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030080644.1704-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-pcm.c | 3 ++- sound/firewire/motu/motu-proc.c | 1 + sound/firewire/motu/motu-protocol-v2.c | 3 +++ sound/firewire/motu/motu-protocol-v3.c | 2 ++ sound/firewire/motu/motu.h | 1 + 5 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 55d3d6661731..490408b49157 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -159,7 +159,8 @@ static int pcm_open(struct snd_pcm_substream *substream) // When source of clock is not internal or any stream is reserved for // transmission of PCM frames, the available sampling rate is limited // at current one. - if (src != SND_MOTU_CLOCK_SOURCE_INTERNAL || + if ((src != SND_MOTU_CLOCK_SOURCE_INTERNAL && + src != SND_MOTU_CLOCK_SOURCE_SPH) || (motu->substreams_counter > 0 && d->events_per_period > 0)) { unsigned int frames_per_period = d->events_per_period; unsigned int frames_per_buffer = d->events_per_buffer; diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index 4e5767c0c8d7..b47ba818343f 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -19,6 +19,7 @@ static const char *const clock_names[] = { [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface", [SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface", [SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface", + [SND_MOTU_CLOCK_SOURCE_SPH] = "Source packet header", [SND_MOTU_CLOCK_SOURCE_UNKNOWN] = "Unknown", }; diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index a42f77fc3858..2757e5b42b9b 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -114,6 +114,9 @@ static int v2_get_clock_source(struct snd_motu *motu, case 2: *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; break; + case 3: + *src = SND_MOTU_CLOCK_SOURCE_SPH; + break; case 4: *src = SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC; break; diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index ba5794c724be..d1545e2b5caa 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -104,6 +104,8 @@ static int v3_get_clock_source(struct snd_motu *motu, *src = SND_MOTU_CLOCK_SOURCE_INTERNAL; } else if (val == 0x01) { *src = SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC; + } else if (val == 0x02) { + *src = SND_MOTU_CLOCK_SOURCE_SPH; } else if (val == 0x10) { *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; } else if (val == 0x18 || val == 0x19) { diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 08e3d568221d..bd782ee4e763 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -104,6 +104,7 @@ enum snd_motu_clock_source { SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX, SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR, SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC, + SND_MOTU_CLOCK_SOURCE_SPH, SND_MOTU_CLOCK_SOURCE_UNKNOWN, }; From 4b2079f80aa065b237955e75d0b627943656b0ab Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Oct 2019 17:06:42 +0900 Subject: [PATCH 049/136] ALSA: firewire-motu: minor code refactoring for protocol version 2 This commit adds some helper functions to parse register value for source of sampling clock and nominal sampling transmission frequency. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030080644.1704-5-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 65 +++++++++++++++----------- 1 file changed, 39 insertions(+), 26 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index 2757e5b42b9b..cb0e58a9f33d 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -26,10 +26,20 @@ #define V2_OPT_IFACE_MODE_ADAT 1 #define V2_OPT_IFACE_MODE_SPDIF 2 +static int get_clock_rate(u32 data, unsigned int *rate) +{ + unsigned int index = (data & V2_CLOCK_RATE_MASK) >> V2_CLOCK_RATE_SHIFT; + if (index >= ARRAY_SIZE(snd_motu_clock_rates)) + return -EIO; + + *rate = snd_motu_clock_rates[index]; + + return 0; +} + static int v2_get_clock_rate(struct snd_motu *motu, unsigned int *rate) { __be32 reg; - unsigned int index; int err; err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, ®, @@ -37,13 +47,7 @@ static int v2_get_clock_rate(struct snd_motu *motu, unsigned int *rate) if (err < 0) return err; - index = (be32_to_cpu(reg) & V2_CLOCK_RATE_MASK) >> V2_CLOCK_RATE_SHIFT; - if (index >= ARRAY_SIZE(snd_motu_clock_rates)) - return -EIO; - - *rate = snd_motu_clock_rates[index]; - - return 0; + return get_clock_rate(be32_to_cpu(reg), rate); } static int v2_set_clock_rate(struct snd_motu *motu, unsigned int rate) @@ -79,38 +83,33 @@ static int v2_set_clock_rate(struct snd_motu *motu, unsigned int rate) sizeof(reg)); } -static int v2_get_clock_source(struct snd_motu *motu, - enum snd_motu_clock_source *src) +static int get_clock_source(struct snd_motu *motu, u32 data, + enum snd_motu_clock_source *src) { - __be32 reg; - unsigned int index; - int err; - - err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, ®, - sizeof(reg)); - if (err < 0) - return err; - - index = be32_to_cpu(reg) & V2_CLOCK_SRC_MASK; + unsigned int index = data & V2_CLOCK_SRC_MASK; if (index > 5) return -EIO; - /* To check the configuration of optical interface. */ - err = snd_motu_transaction_read(motu, V2_IN_OUT_CONF_OFFSET, ®, - sizeof(reg)); - if (err < 0) - return err; - switch (index) { case 0: *src = SND_MOTU_CLOCK_SOURCE_INTERNAL; break; case 1: + { + __be32 reg; + + // To check the configuration of optical interface. + int err = snd_motu_transaction_read(motu, V2_IN_OUT_CONF_OFFSET, + ®, sizeof(reg)); + if (err < 0) + return err; + if (be32_to_cpu(reg) & 0x00000200) *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT; else *src = SND_MOTU_CLOCK_SOURCE_ADAT_ON_OPT; break; + } case 2: *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; break; @@ -130,6 +129,20 @@ static int v2_get_clock_source(struct snd_motu *motu, return 0; } +static int v2_get_clock_source(struct snd_motu *motu, + enum snd_motu_clock_source *src) +{ + __be32 reg; + int err; + + err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, ®, + sizeof(reg)); + if (err < 0) + return err; + + return get_clock_source(motu, be32_to_cpu(reg), src); +} + static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable) { __be32 reg; From bd1073728260e1d2f9589c928b6d8ddb14f5646f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Oct 2019 17:06:43 +0900 Subject: [PATCH 050/136] ALSA: firewire-motu: code refactoring to handle model specific switch for protocol v2 In MOTU FireWire series, devices which support protocol version 2 have several types of hardware design to process audio data frames for isoc packet. Roughly devices are categorized into three groups: - 828mkII - Traveler/896HD - UltraLite/8pre FireWire Some bit flags in register addressed by 0x'ffff'f000'0b14 includes device-specific effects. This commit cleanups implementation of protocol v2 in this point. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030080644.1704-6-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-protocol-v2.c | 72 +++++++++++++------------- sound/firewire/motu/motu.c | 12 ++--- sound/firewire/motu/motu.h | 3 +- 3 files changed, 43 insertions(+), 44 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v2.c b/sound/firewire/motu/motu-protocol-v2.c index cb0e58a9f33d..619b6ae73f62 100644 --- a/sound/firewire/motu/motu-protocol-v2.c +++ b/sound/firewire/motu/motu-protocol-v2.c @@ -12,10 +12,8 @@ #define V2_CLOCK_RATE_SHIFT 3 #define V2_CLOCK_SRC_MASK 0x00000007 #define V2_CLOCK_SRC_SHIFT 0 -#define V2_CLOCK_TRAVELER_FETCH_DISABLE 0x04000000 -#define V2_CLOCK_TRAVELER_FETCH_ENABLE 0x03000000 -#define V2_CLOCK_8PRE_FETCH_DISABLE 0x02000000 -#define V2_CLOCK_8PRE_FETCH_ENABLE 0x00000000 +#define V2_CLOCK_FETCH_ENABLE 0x02000000 +#define V2_CLOCK_MODEL_SPECIFIC 0x04000000 #define V2_IN_OUT_CONF_OFFSET 0x0c04 #define V2_OPT_OUT_IFACE_MASK 0x00000c00 @@ -73,11 +71,6 @@ static int v2_set_clock_rate(struct snd_motu *motu, unsigned int rate) data &= ~V2_CLOCK_RATE_MASK; data |= i << V2_CLOCK_RATE_SHIFT; - if (motu->spec == &snd_motu_spec_traveler) { - data &= ~V2_CLOCK_TRAVELER_FETCH_ENABLE; - data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; - } - reg = cpu_to_be32(data); return snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, ®, sizeof(reg)); @@ -145,42 +138,49 @@ static int v2_get_clock_source(struct snd_motu *motu, static int v2_switch_fetching_mode(struct snd_motu *motu, bool enable) { + enum snd_motu_clock_source src; __be32 reg; u32 data; int err = 0; - if (motu->spec == &snd_motu_spec_traveler || - motu->spec == &snd_motu_spec_8pre) { - err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, - ®, sizeof(reg)); + // 828mkII implements Altera ACEX 1K EP1K30. Nothing to do. + if (motu->spec == &snd_motu_spec_828mk2) + return 0; + + err = snd_motu_transaction_read(motu, V2_CLOCK_STATUS_OFFSET, ®, + sizeof(reg)); + if (err < 0) + return err; + data = be32_to_cpu(reg); + + err = get_clock_source(motu, data, &src); + if (err < 0) + return err; + + data &= ~(V2_CLOCK_FETCH_ENABLE | V2_CLOCK_MODEL_SPECIFIC); + if (enable) + data |= V2_CLOCK_FETCH_ENABLE; + + if (motu->spec->flags & SND_MOTU_SPEC_SUPPORT_CLOCK_X4) { + // Expected for Traveler and 896HD, which implements Altera + // Cyclone EP1C3. + data |= V2_CLOCK_MODEL_SPECIFIC; + } else { + // For UltraLite and 8pre, which implements Xilinx Spartan + // XC3S200. + unsigned int rate; + + err = get_clock_rate(data, &rate); if (err < 0) return err; - data = be32_to_cpu(reg); - if (motu->spec == &snd_motu_spec_traveler) { - data &= ~(V2_CLOCK_TRAVELER_FETCH_DISABLE | - V2_CLOCK_TRAVELER_FETCH_ENABLE); - - if (enable) - data |= V2_CLOCK_TRAVELER_FETCH_ENABLE; - else - data |= V2_CLOCK_TRAVELER_FETCH_DISABLE; - } else if (motu->spec == &snd_motu_spec_8pre) { - data &= ~(V2_CLOCK_8PRE_FETCH_DISABLE | - V2_CLOCK_8PRE_FETCH_ENABLE); - - if (enable) - data |= V2_CLOCK_8PRE_FETCH_DISABLE; - else - data |= V2_CLOCK_8PRE_FETCH_ENABLE; - } - - reg = cpu_to_be32(data); - err = snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, - ®, sizeof(reg)); + if (src == SND_MOTU_CLOCK_SOURCE_SPH && rate > 48000) + data |= V2_CLOCK_MODEL_SPECIFIC; } - return err; + reg = cpu_to_be32(data); + return snd_motu_transaction_write(motu, V2_CLOCK_STATUS_OFFSET, ®, + sizeof(reg)); } static void calculate_fixed_part(struct snd_motu_packet_format *formats, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 680e363d0a4b..f249ad39d8a8 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -172,7 +172,7 @@ static void motu_bus_update(struct fw_unit *unit) snd_motu_transaction_reregister(motu); } -static const struct snd_motu_spec motu_828mk2 = { +const struct snd_motu_spec snd_motu_spec_828mk2 = { .name = "828mk2", .protocol = &snd_motu_protocol_v2, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | @@ -187,7 +187,7 @@ static const struct snd_motu_spec motu_828mk2 = { .analog_out_ports = 8, }; -const struct snd_motu_spec snd_motu_spec_traveler = { +static const struct snd_motu_spec motu_traveler = { .name = "Traveler", .protocol = &snd_motu_protocol_v2, .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | @@ -202,7 +202,7 @@ const struct snd_motu_spec snd_motu_spec_traveler = { .analog_out_ports = 8, }; -const struct snd_motu_spec snd_motu_spec_8pre = { +static const struct snd_motu_spec motu_8pre = { .name = "8pre", .protocol = &snd_motu_protocol_v2, // In tx, use coax chunks for mix-return 1/2. In rx, use coax chunks for @@ -270,9 +270,9 @@ static const struct snd_motu_spec motu_4pre = { } static const struct ieee1394_device_id motu_id_table[] = { - SND_MOTU_DEV_ENTRY(0x000003, &motu_828mk2), - SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), - SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), + SND_MOTU_DEV_ENTRY(0x000003, &snd_motu_spec_828mk2), + SND_MOTU_DEV_ENTRY(0x000009, &motu_traveler), + SND_MOTU_DEV_ENTRY(0x00000f, &motu_8pre), SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index bd782ee4e763..6efbde405a0d 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -130,8 +130,7 @@ struct snd_motu_spec { extern const struct snd_motu_protocol snd_motu_protocol_v2; extern const struct snd_motu_protocol snd_motu_protocol_v3; -extern const struct snd_motu_spec snd_motu_spec_traveler; -extern const struct snd_motu_spec snd_motu_spec_8pre; +extern const struct snd_motu_spec snd_motu_spec_828mk2; int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, enum amdtp_stream_direction dir, From 271e657f92ed94ced8a9fccf265b993650272a6b Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Oct 2019 17:06:44 +0900 Subject: [PATCH 051/136] ALSA: firewire-motu: add support for MOTU UltraLite UltraLite was shipped in 2005 by MOTU, and already discontinued. This model consists of below ICs: - Texus Instruments TSB41AB2 for physical layer of IEEE 1394 bus - Xilinx Spartan XC35S200 for link layer of IEEE 1394 bus, protocol layer and signal processing This commit adds support for this model. Like the other MOTU models, ALSA firewire MOTU driver fails to drive the device for stable sampling clock and generate noisy sound. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04107574 bus_info_length 4, crc_length 16, crc 30068 404 31333934 bus_name "1394" 408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4) 40c 0001f200 company_id 0001f2 | 410 0007b82d device_id 000007b82d | EUI-64 0001f2000007b82d root directory ----------------------------------------------------------------- 414 0004c65c directory_length 4, crc 50780 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 8d000006 --> eui-64 leaf at 438 424 d1000001 --> unit directory at 428 unit directory at 428 ----------------------------------------------------------------- 428 0003d80a directory_length 3, crc 55306 42c 120001f2 specifier id 430 1300000d version 434 17100800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 00024566 leaf_length 2, crc 17766 43c 0001f200 company_id 0001f2 | 440 0007b82d device_id 000007b82d | EUI-64 0001f2000007b82d Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030080644.1704-7-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index f249ad39d8a8..f2080d720aa9 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -202,6 +202,19 @@ static const struct snd_motu_spec motu_traveler = { .analog_out_ports = 8, }; +static const struct snd_motu_spec motu_ultralite = { + .name = "UltraLite", + .protocol = &snd_motu_protocol_v2, + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_TX_MICINST_CHUNK | // padding. + SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_MIDI_2ND_Q | + SND_MOTU_SPEC_TX_MIDI_2ND_Q | + SND_MOTU_SPEC_RX_SEPARATED_MAIN, + .analog_in_ports = 8, + .analog_out_ports = 8, +}; + static const struct snd_motu_spec motu_8pre = { .name = "8pre", .protocol = &snd_motu_protocol_v2, @@ -272,6 +285,7 @@ static const struct snd_motu_spec motu_4pre = { static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000003, &snd_motu_spec_828mk2), SND_MOTU_DEV_ENTRY(0x000009, &motu_traveler), + SND_MOTU_DEV_ENTRY(0x00000d, &motu_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &motu_8pre), SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ From 0929249e3be3bb82ee6cfec0025f4dde952210b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 30 Oct 2019 11:09:21 +0100 Subject: [PATCH 052/136] ALSA: firewire-motu: Correct a typo in the clock proc string Just fix a typo of "S/PDIF" in the clock name string. Fixes: 4638ec6ede08 ("ALSA: firewire-motu: add proc node to show current statuc of clock and packet formats") Acked-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191030100921.3826-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/firewire/motu/motu-proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index b47ba818343f..187f6abd878c 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -16,7 +16,7 @@ static const char *const clock_names[] = { [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT] = "S/PDIF on optical interface", [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_A] = "S/PDIF on optical interface A", [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_B] = "S/PDIF on optical interface B", - [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface", + [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PDIF on coaxial interface", [SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface", [SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface", [SND_MOTU_CLOCK_SOURCE_SPH] = "Source packet header", From d3eabe939aee3ffd5b133766a932629a9746298c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 1 Nov 2019 22:13:21 +0900 Subject: [PATCH 053/136] ALSA: bebob: expand sleep just after breaking connections for protocol version 1 As long as I investigated, some devices with BeBoB protocol version 1 can be freezed during several hundreds milliseconds after breaking connections. When accessing during the freezed time, any transaction is corrupted. In the worst case, the device is going to reboot. I can see this issue in: * Roland FA-66 * M-Audio FireWire Solo This commit expands sleep just after breaking connections to avoid the freezed time as much as possible. I note that the freeze/reboot behaviour is similar to below models: * Focusrite Saffire Pro 10 I/O * Focusrite Saffire Pro 26 I/O The above models certainly reboot after breaking connections. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191101131323.17300-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_stream.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index f7f0db5aa811..1b264d0d63eb 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -415,15 +415,16 @@ static int make_both_connections(struct snd_bebob *bebob) return 0; } -static void -break_both_connections(struct snd_bebob *bebob) +static void break_both_connections(struct snd_bebob *bebob) { cmp_connection_break(&bebob->in_conn); cmp_connection_break(&bebob->out_conn); - /* These models seems to be in transition state for a longer time. */ - if (bebob->maudio_special_quirk != NULL) - msleep(200); + // These models seem to be in transition state for a longer time. When + // accessing in the state, any transactions is corrupted. In the worst + // case, the device is going to reboot. + if (bebob->version < 2) + msleep(600); } static int From da5e83374f463a4dd91321428d5ea5230b6d87b4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 1 Nov 2019 22:13:22 +0900 Subject: [PATCH 054/136] ALSA: bebob: expand delay of start for IR context just for version 3 firmware As long as I investigated, there's some cases about the delay for device between establishing OUTPUT_PLUG and transmitting first isoc packet. For devices which support BeBoB protocol version 1 can transmit the packet within several hundred milliseconds, while for devices which support BeBoB protocol version 3 can transmit the packet within 2 seconds. Devices with protocol version 1: * Edirol FA-66 * Yamaha GO46 * Terratec Phase x24 FW * M-Audio FireWire AudioPhile * M-Audio FireWire Solo * M-Audio FireWire 1814 * M-Audio FireWire 410 * Focusrite Saffire Pro 26 I/O Devices with protocol version 3: * M-Audio Profire Lightbridge * Behringer FCA610 * Phonic Firefly 202 At present ALSA bebob driver postpones starting IR context during 1.5 sec for all of supported devices. The delay is too long for devices with protocol version 1, while it's not enough for devices with protocol version 3. This commit improves the delay for these protocols. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191101131323.17300-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_stream.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 1b264d0d63eb..895a113635af 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -7,7 +7,7 @@ #include "./bebob.h" -#define CALLBACK_TIMEOUT 2000 +#define CALLBACK_TIMEOUT 2500 #define FW_ISO_RESOURCE_DELAY 1000 /* @@ -639,6 +639,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) if (!amdtp_stream_running(&bebob->rx_stream)) { unsigned int curr_rate; + unsigned int ir_delay_cycle; if (bebob->maudio_special_quirk) { err = bebob->spec->rate->get(bebob, &curr_rate); @@ -660,13 +661,18 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) // The device postpones start of transmission mostly for 1 sec // after receives packets firstly. For safe, IR context starts - // 1.5 sec (=12000 cycles) later. This is within 2.0 sec - // (=CALLBACK_TIMEOUT). + // 0.4 sec (=3200 cycles) later to version 1 or 2 firmware, + // 2.0 sec (=16000 cycles) for version 3 firmware. This is + // within 2.5 sec (=CALLBACK_TIMEOUT). // Furthermore, some devices transfer isoc packets with // discontinuous counter in the beginning of packet streaming. // The delay has an effect to avoid detection of this // discontinuity. - err = amdtp_domain_start(&bebob->domain, 12000); + if (bebob->version < 2) + ir_delay_cycle = 3200; + else + ir_delay_cycle = 16000; + err = amdtp_domain_start(&bebob->domain, ir_delay_cycle); if (err < 0) goto error; From 6c94f380c6c1d0206da55b0342557926052f2b3e Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 1 Nov 2019 22:13:23 +0900 Subject: [PATCH 055/136] ALSA: bebob: link the order of establishing connections and Syt-match clock mode Originally BeBeB ASICs and firmware supports clock mode to synchronizing to syt field of received isoc packet. This mode is known as 'SYT Match' slightly described in IEC 61883-6 (but no detail mechanisms). In this mode, drivers can control sampling clock in device. Driver for Windows and macOS uses this feature to perform synchronization for devices on the same bus. In this mode, a plug of Music subunit for synchronization is connected to a plug of isoc unit for incoming packet streaming, then the order to establish connections is INPUT_PLUG first, OUTPUT_PLUG second. This commit implements the above. Actually each device works with its own clock for sampling, therefore the original design is hardly implemented to vendor's products. Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191101131323.17300-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_stream.c | 51 +++++++++++++---------------- 1 file changed, 22 insertions(+), 29 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 895a113635af..bbae04793c50 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -398,23 +398,6 @@ check_connection_used_by_others(struct snd_bebob *bebob, struct amdtp_stream *s) return err; } -static int make_both_connections(struct snd_bebob *bebob) -{ - int err = 0; - - err = cmp_connection_establish(&bebob->out_conn); - if (err < 0) - return err; - - err = cmp_connection_establish(&bebob->in_conn); - if (err < 0) { - cmp_connection_break(&bebob->out_conn); - return err; - } - - return 0; -} - static void break_both_connections(struct snd_bebob *bebob) { cmp_connection_break(&bebob->in_conn); @@ -427,8 +410,7 @@ static void break_both_connections(struct snd_bebob *bebob) msleep(600); } -static int -start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) +static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) { struct cmp_connection *conn; int err = 0; @@ -438,18 +420,19 @@ start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) else conn = &bebob->out_conn; - /* channel mapping */ + // channel mapping. if (bebob->maudio_special_quirk == NULL) { err = map_data_channels(bebob, stream); if (err < 0) - goto end; + return err; } - // start amdtp stream. - err = amdtp_domain_add_stream(&bebob->domain, stream, - conn->resources.channel, conn->speed); -end: - return err; + err = cmp_connection_establish(conn); + if (err < 0) + return err; + + return amdtp_domain_add_stream(&bebob->domain, stream, + conn->resources.channel, conn->speed); } static int init_stream(struct snd_bebob *bebob, struct amdtp_stream *stream) @@ -638,6 +621,8 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) } if (!amdtp_stream_running(&bebob->rx_stream)) { + enum snd_bebob_clock_type src; + struct amdtp_stream *master, *slave; unsigned int curr_rate; unsigned int ir_delay_cycle; @@ -647,15 +632,23 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob) return err; } - err = make_both_connections(bebob); + err = snd_bebob_stream_get_clock_src(bebob, &src); if (err < 0) return err; - err = start_stream(bebob, &bebob->rx_stream); + if (src != SND_BEBOB_CLOCK_TYPE_SYT) { + master = &bebob->tx_stream; + slave = &bebob->rx_stream; + } else { + master = &bebob->rx_stream; + slave = &bebob->tx_stream; + } + + err = start_stream(bebob, master); if (err < 0) goto error; - err = start_stream(bebob, &bebob->tx_stream); + err = start_stream(bebob, slave); if (err < 0) goto error; From 6111fd2370eecae9f11bfdc08ba097e0b51fcfd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2019 11:11:15 +0100 Subject: [PATCH 056/136] ALSA: pcm: Fix missing check of the new non-cached buffer type The check for the mmap support via hw_support_mmap() function misses the case where the device is with SNDRV_DMA_TYPE_DEV_UC, which should have been treated equally as SNDRV_DMA_TYPE_DEV. Let's fix it. Note that this bug doesn't hit any practical problem, because SNDRV_DMA_TYPE_DEV_UC is used only for x86-specific drivers (snd-hda-intel and snd-intel8x0) for the specific platforms that need the non-cached buffers. And, on such platforms, hw_support_mmap() already returns true in anyway. That's the reason I didn't put Cc-to-stable mark here. This is only for any theoretical future extension. Fixes: 425da159707b ("ALSA: pcm: use dma_can_mmap() to check if a device supports dma_mmap_*") Fixes: 42e748a0b325 ("ALSA: memalloc: Add non-cached buffer type") Link: https://lore.kernel.org/r/20191104101115.27311-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 91c6ad58729f..c3a139436ac2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -222,7 +222,8 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) return false; if (substream->ops->mmap || - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV) + (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && + substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) return true; return dma_can_mmap(substream->dma_buffer.dev.dev); From 42ec336f1f9d54049811b749f729e9e01c152ade Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 09:18:06 +0100 Subject: [PATCH 057/136] ALSA: hda: Disable regmap internal locking Since we apply the own mutex (bus->cmd_mutex) in HDA core side, the internal locking in regmap is superfluous. This patch adds the flag to indicate that. Also, an infamous side-effect by this change is that it disables the regmap debugfs, too, and this is seen rather good; the regmap debugfs isn't quite useful for HD-audio as it provides the very sparse registers and its debugfs access tends to lead to the way too high resource usages or sometimes hang up. So it'd be rather safe to disable it altogether. Link: https://lore.kernel.org/r/2029139028.10333037.1572874551626.JavaMail.zimbra@redhat.com Link: https://lore.kernel.org/r/20191105081806.4896-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_regmap.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 286361ecd640..906b1e20bae0 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,6 +363,7 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, + .disable_locking = true, }; /** From 08422d2c559d1119b17c14e79b2d8ff89f46a356 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 09:01:35 +0100 Subject: [PATCH 058/136] ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type Currently we pass the artificial device pointer to the allocation helper in the case of SNDRV_DMA_TYPE_CONTINUOUS for passing the GFP flags. But all common cases are the allocations with GFP_KERNEL, and it's messy to put this in each place. In this patch, the memalloc core helper is changed to accept the NULL device pointer and it treats as the default mode, GFP_KERNEL, so that all callers can omit the complex argument but just leave NULL. Link: https://lore.kernel.org/r/20191105080138.1260-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- .../sound/kernel-api/writing-an-alsa-driver.rst | 14 ++++++++------ sound/core/memalloc.c | 11 ++++++++--- 2 files changed, 16 insertions(+), 9 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 132f5eb9b530..5385618fd881 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3523,12 +3523,14 @@ The second argument (type) and the third argument (device pointer) are dependent on the bus. For normal devices, pass the device pointer (typically identical as ``card->dev``) to the third argument with ``SNDRV_DMA_TYPE_DEV`` type. For the continuous buffer unrelated to the -bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type and the -``snd_dma_continuous_data(GFP_KERNEL)`` device pointer, where -``GFP_KERNEL`` is the kernel allocation flag to use. For the -scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the device -pointer (see the `Non-Contiguous Buffers`_ -section). +bus can be pre-allocated with ``SNDRV_DMA_TYPE_CONTINUOUS`` type. +You can pass NULL to the device pointer in that case, which is the +default mode implying to allocate with ``GFP_KRENEL`` flag. +If you need a different GFP flag, you can pass it by encoding the flag +into the device pointer via a special macro +:c:func:`snd_dma_continuous_data()`. +For the scatter-gather buffers, use ``SNDRV_DMA_TYPE_DEV_SG`` with the +device pointer (see the `Non-Contiguous Buffers`_ section). Once the buffer is pre-allocated, you can use the allocator in the ``hw_params`` callback: diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6850d13aa98c..1b1c7620cbda 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -99,6 +99,13 @@ static void snd_free_dev_iram(struct snd_dma_buffer *dmab) * */ +static inline gfp_t snd_mem_get_gfp_flags(const struct device *dev) +{ + if (!dev) + return GFP_KERNEL; + else + return (__force gfp_t)(unsigned long)dev; +} /** * snd_dma_alloc_pages - allocate the buffer area according to the given type @@ -120,8 +127,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, return -ENXIO; if (WARN_ON(!dmab)) return -ENXIO; - if (WARN_ON(!device)) - return -EINVAL; dmab->dev.type = type; dmab->dev.dev = device; @@ -129,7 +134,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, switch (type) { case SNDRV_DMA_TYPE_CONTINUOUS: dmab->area = alloc_pages_exact(size, - (__force gfp_t)(unsigned long)device); + snd_mem_get_gfp_flags(device)); dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA From 1fe7f397cfe229494a88c6496b53ba532d4d4a42 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 09:01:36 +0100 Subject: [PATCH 059/136] ALSA: memalloc: Add vmalloc buffer allocation support This patch adds the vmalloc buffer support to ALSA memalloc core. A new type, SNDRV_DMA_TYPE_VMALLOC was added. The vmalloc buffer has been already supported in the PCM via a few own helper functions, but the user sometimes get confused and misuse them. With this patch, the whole buffer management is integrated into the memalloc core, so they can be used in a sole common way. Link: https://lore.kernel.org/r/20191105080138.1260-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 20 ++++++++++++++++---- 2 files changed, 17 insertions(+), 4 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 240622d89c0b..6ada3b8ede4e 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -44,6 +44,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_IRAM SNDRV_DMA_TYPE_DEV #endif +#define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 1b1c7620cbda..a83553fbedf0 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -10,6 +10,7 @@ #include #include #include +#include #ifdef CONFIG_X86 #include #endif @@ -99,10 +100,11 @@ static void snd_free_dev_iram(struct snd_dma_buffer *dmab) * */ -static inline gfp_t snd_mem_get_gfp_flags(const struct device *dev) +static inline gfp_t snd_mem_get_gfp_flags(const struct device *dev, + gfp_t default_gfp) { if (!dev) - return GFP_KERNEL; + return default_gfp; else return (__force gfp_t)(unsigned long)dev; } @@ -123,6 +125,8 @@ static inline gfp_t snd_mem_get_gfp_flags(const struct device *dev) int snd_dma_alloc_pages(int type, struct device *device, size_t size, struct snd_dma_buffer *dmab) { + gfp_t gfp; + if (WARN_ON(!size)) return -ENXIO; if (WARN_ON(!dmab)) @@ -133,8 +137,13 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->bytes = 0; switch (type) { case SNDRV_DMA_TYPE_CONTINUOUS: - dmab->area = alloc_pages_exact(size, - snd_mem_get_gfp_flags(device)); + gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL); + dmab->area = alloc_pages_exact(size, gfp); + dmab->addr = 0; + break; + case SNDRV_DMA_TYPE_VMALLOC: + gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL | __GFP_HIGHMEM); + dmab->area = __vmalloc(size, gfp, PAGE_KERNEL); dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA @@ -220,6 +229,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_CONTINUOUS: free_pages_exact(dmab->area, dmab->bytes); break; + case SNDRV_DMA_TYPE_VMALLOC: + vfree(dmab->area); + break; #ifdef CONFIG_HAS_DMA #ifdef CONFIG_GENERIC_ALLOCATOR case SNDRV_DMA_TYPE_DEV_IRAM: From 7e8edae39fd1f880f098612c991cf6d0f646c20d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 09:01:37 +0100 Subject: [PATCH 060/136] ALSA: pcm: Handle special page mapping in the default mmap handler When a driver needs to deal with a special buffer like a SG or a vmalloc buffer, it has to set up the PCM page ops explicitly for the corresponding helper function. This is rather error-prone and many people forgot or incorrectly used it. For simplifying the call patterns and avoiding such a potential bug, this patch enhances the PCM default mmap handler to check the (pre-)allocated buffer type and handles the page gracefully depending on the buffer type. If the PCM page ops is given, the ops is still used in a higher priority. The new code path is only for the default (NULL page ops) case. Link: https://lore.kernel.org/r/20191105080138.1260-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c3a139436ac2..998c63192ae4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -3335,7 +3336,18 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; - return virt_to_page(vaddr); + + switch (substream->dma_buffer.dev.type) { +#ifdef CONFIG_SND_DMA_SGBUF + case SNDRV_DMA_TYPE_DEV_SG: + case SNDRV_DMA_TYPE_DEV_UC_SG: + return snd_pcm_sgbuf_ops_page(substream, ofs); +#endif /* CONFIG_SND_DMA_SGBUF */ + case SNDRV_DMA_TYPE_VMALLOC: + return vmalloc_to_page(vaddr); + default: + return virt_to_page(vaddr); + } } /* From abffd8d0d5380e2c37759ba6be1f6bf47bfe975f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 09:01:38 +0100 Subject: [PATCH 061/136] ALSA: docs: Update documentation about SG- and vmalloc-buffers The recent changes simplified the required setup for SG- and vmalloc- buffers. Update the documentation accordingly. Link: https://lore.kernel.org/r/20191105080138.1260-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- .../kernel-api/writing-an-alsa-driver.rst | 68 +++++++++++-------- 1 file changed, 38 insertions(+), 30 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 5385618fd881..ba008ce28029 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -2095,10 +2095,12 @@ This callback is atomic as default. page callback ~~~~~~~~~~~~~ -This callback is optional too. This callback is used mainly for -non-contiguous buffers. The mmap calls this callback to get the page -address. Some examples will be explained in the later section `Buffer -and Memory Management`_, too. +This callback is optional too. The mmap calls this callback to get the +page fault address. + +Since the recent changes, you need no special callback any longer for +the standard SG-buffer or vmalloc-buffer. Hence this callback should +be rarely used. mmap calllback ~~~~~~~~~~~~~~ @@ -3700,8 +3702,15 @@ For creating the SG-buffer handler, call ``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI pre-allocator. You need to pass ``snd_dma_pci_data(pci)``, where pci is the :c:type:`struct pci_dev ` pointer of the chip as -well. The ``struct snd_sg_buf`` instance is created as -``substream->dma_private``. You can cast the pointer like: +well. + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + snd_dma_pci_data(pci), size, max); + +The ``struct snd_sg_buf`` instance is created as +``substream->dma_private`` in turn. You can cast the pointer like: :: @@ -3717,10 +3726,6 @@ physically non-contiguous. The physical address table is set up in ``sgbuf->table``. You can get the physical address at a certain offset via :c:func:`snd_pcm_sgbuf_get_addr()`. -When a SG-handler is used, you need to set -:c:func:`snd_pcm_sgbuf_ops_page()` as the ``page`` callback. (See -`page callback`_ section.) - To release the data, call :c:func:`snd_pcm_lib_free_pages()` in the ``hw_free`` callback as usual. @@ -3728,30 +3733,33 @@ Vmalloc'ed Buffers ------------------ It's possible to use a buffer allocated via :c:func:`vmalloc()`, for -example, for an intermediate buffer. Since the allocated pages are not -contiguous, you need to set the ``page`` callback to obtain the physical -address at every offset. - -The easiest way to achieve it would be to use -:c:func:`snd_pcm_lib_alloc_vmalloc_buffer()` for allocating the buffer -via :c:func:`vmalloc()`, and set :c:func:`snd_pcm_sgbuf_ops_page()` to -the ``page`` callback. At release, you need to call -:c:func:`snd_pcm_lib_free_vmalloc_buffer()`. - -If you want to implementation the ``page`` manually, it would be like -this: +example, for an intermediate buffer. In the recent version of kernel, +you can simply allocate it via standard +:c:func:`snd_pcm_lib_malloc_pages()` and co after setting up the +buffer preallocation with ``SNDRV_DMA_TYPE_VMALLOC`` type. :: - #include + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); - /* get the physical page pointer on the given offset */ - static struct page *mychip_page(struct snd_pcm_substream *substream, - unsigned long offset) - { - void *pageptr = substream->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); - } +The NULL is passed to the device pointer argument, which indicates +that the default pages (GFP_KERNEL and GFP_HIGHMEM) will be +allocated. + +Also, note that zero is passed to both the size and the max size +arguments here. Since each vmalloc call should succeed at any time, +we don't need to pre-allocate the buffers like other continuous +pages. + +If you need the 32bit DMA allocation, pass the device pointer encoded +by :c:func:`snd_dma_continuous_data()` with ``GFP_KERNEL|__GFP_DMA32`` +argument. + +:: + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + snd_dma_continuous_data(GFP_KERNEL | __GFP_DMA32), 0, 0); Proc Interface ============== From a55eaf177a36f7789eec0985a7edd00fe7353d90 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 20:10:06 +0100 Subject: [PATCH 062/136] ALSA: pcm: Warn if doubly preallocated Warn if snd_pcm_lib_preallocate_pages*() is applied to the stream that has already the preallocated buffers and skip the allocation. It's a clearly a driver bug. Link: https://lore.kernel.org/r/20191105191007.18150-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 7600dcdf5fd4..19407f79b638 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -221,6 +221,8 @@ void snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, int type, struct device *data, size_t size, size_t max) { + if (snd_BUG_ON(substream->dma_buffer.dev.type)) + return; substream->dma_buffer.dev.type = type; substream->dma_buffer.dev.dev = data; snd_pcm_lib_preallocate_pages1(substream, size, max); From d39789912f150b66515ccda7ed00bba828a334d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 20:10:07 +0100 Subject: [PATCH 063/136] ALSA: pcm: Create proc files only for non-empty preallocations It makes little sense to create prealloc proc files for streams that have the zero max size, which is a typical case for vmalloc buffers. Skip the proc file creations to save resources in such a case. Link: https://lore.kernel.org/r/20191105191007.18150-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 19407f79b638..17ee361ce791 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -203,7 +203,8 @@ static void snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream, if (substream->dma_buffer.bytes > 0) substream->buffer_bytes_max = substream->dma_buffer.bytes; substream->dma_max = max; - preallocate_info_init(substream); + if (max > 0) + preallocate_info_init(substream); } From 8fd9da750e76c321cd0a781c89a0d70cc399b22d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:33 +0100 Subject: [PATCH 064/136] ALSA: drivers: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Link: https://lore.kernel.org/r/20191105151856.10785-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 2 +- sound/drivers/ml403-ac97cr.c | 2 +- sound/drivers/pcsp/pcsp_lib.c | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index aee7c04d49e5..022a0db692e0 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -702,7 +702,7 @@ static int snd_card_dummy_pcm(struct snd_dummy *dummy, int device, if (!fake_buffer) { snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 0, 64*1024); } return 0; diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index a3c1c064d1b5..70a6d1832698 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1242,7 +1242,7 @@ snd_ml403_ac97cr_pcm(struct snd_ml403_ac97cr *ml403_ac97cr, int device) ml403_ac97cr->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64 * 1024, 128 * 1024); return 0; diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 8f0f05bbc081..f91316bf01cb 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -352,8 +352,8 @@ int snd_pcsp_new_pcm(struct snd_pcsp *chip) snd_pcm_lib_preallocate_pages_for_all(chip->pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data - (GFP_KERNEL), PCSP_BUFFER_SIZE, + NULL, + PCSP_BUFFER_SIZE, PCSP_BUFFER_SIZE); return 0; From 987334266d087a4974c73f4a071bc24596f545aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:34 +0100 Subject: [PATCH 065/136] ALSA: mips: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Link: https://lore.kernel.org/r/20191105151856.10785-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/mips/hal2.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 6676bcbd769f..c9e060939708 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -741,8 +741,7 @@ static int hal2_pcm_create(struct snd_hal2 *hal2) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &hal2_capture_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - 0, 1024 * 1024); + NULL, 0, 1024 * 1024); return 0; } From 1422bc5001a98f5be618a4abc62325f8a9310941 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:35 +0100 Subject: [PATCH 066/136] ALSA: rme32: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Link: https://lore.kernel.org/r/20191105151856.10785-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/rme32.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 40cc6ca88f7b..58a4b8df25d4 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1375,7 +1375,7 @@ static int snd_rme32_create(struct rme32 *rme32) snd_pcm_set_ops(rme32->spdif_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme32_capture_spdif_fd_ops); snd_pcm_lib_preallocate_pages_for_all(rme32->spdif_pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 0, RME32_MID_BUFFER_SIZE); rme32->spdif_pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; } else { @@ -1407,7 +1407,7 @@ static int snd_rme32_create(struct rme32 *rme32) snd_pcm_set_ops(rme32->adat_pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_rme32_capture_adat_fd_ops); snd_pcm_lib_preallocate_pages_for_all(rme32->adat_pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 0, RME32_MID_BUFFER_SIZE); rme32->adat_pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; } else { From a1aef4881cf1781e3d5c5c07df1b8774d3fe2def Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:36 +0100 Subject: [PATCH 067/136] ALSA: sh: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Link: https://lore.kernel.org/r/20191105151856.10785-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 2 +- sound/sh/sh_dac_audio.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 52e9cfb4f819..bf1fb0d8a930 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -443,7 +443,7 @@ static int __init snd_aicapcmchip(struct snd_card_aica /* Allocate the DMA buffers */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, AICA_BUFFER_SIZE, AICA_BUFFER_SIZE); return 0; diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index ed877a138965..f9e36abc98ac 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -268,7 +268,7 @@ static int snd_sh_dac_pcm(struct snd_sh_dac *chip, int device) /* buffer size=48K */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 48 * 1024, 48 * 1024); From 0382e4e18fafcaea75ffab85838ef2110fac1668 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:37 +0100 Subject: [PATCH 068/136] ALSA: sparc: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Link: https://lore.kernel.org/r/20191105151856.10785-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/sparc/amd7930.c | 2 +- sound/sparc/dbri.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 441222c8e223..d4b8ccc61dc2 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -777,7 +777,7 @@ static int snd_amd7930_pcm(struct snd_amd7930 *amd) amd->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64*1024, 64*1024); return 0; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 6e065d44060e..4911103421ff 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2249,7 +2249,7 @@ static int snd_dbri_pcm(struct snd_card *card) strcpy(pcm->name, card->shortname); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64 * 1024, 64 * 1024); return 0; } From 0af0a4fec0c48d591a3788cb6c08b21a17201dd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:38 +0100 Subject: [PATCH 069/136] ALSA: usb: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Link: https://lore.kernel.org/r/20191105151856.10785-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/line6/pcm.c | 4 +--- sound/usb/usx2y/usbusx2yaudio.c | 4 ++-- sound/usb/usx2y/usx2yhwdeppcm.c | 4 ++-- 3 files changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index f70211e6b174..9c437c716cfd 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -502,9 +502,7 @@ static int snd_line6_new_pcm(struct usb_line6 *line6, struct snd_pcm **pcm_ret) /* pre-allocation of buffers */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data - (GFP_KERNEL), 64 * 1024, - 128 * 1024); + NULL, 64 * 1024, 128 * 1024); return 0; } diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 89fa287678fc..25a0939f410a 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -970,13 +970,13 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, if (playback_endpoint) { snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64*1024, 128*1024); } snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64*1024, 128*1024); usX2Y(card)->pcm_devs++; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index ac8960b6b299..997493e839ee 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -728,11 +728,11 @@ int usX2Y_hwdep_pcm_new(struct snd_card *card) sprintf(pcm->name, NAME_ALLCAPS" hwdep Audio"); snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64*1024, 128*1024); snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, 64*1024, 128*1024); return 0; From 1158e89c9d8f5e676870ab0a729de16ac39e59af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:39 +0100 Subject: [PATCH 070/136] ALSA: 6fire: Drop the dead code A few error handling code was forgotten where it never reaches. Drop it. Link: https://lore.kernel.org/r/20191105151856.10785-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 88ac1c4ee163..4ded944e5d5d 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -660,13 +660,6 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); - if (ret) { - usb6fire_pcm_buffers_destroy(rt); - kfree(rt); - dev_err(&chip->dev->dev, - "error preallocating pcm buffers.\n"); - return ret; - } rt->instance = pcm; chip->pcm = rt; From b315997d7c2295b2b32cb661a584208ab05ffaea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:40 +0100 Subject: [PATCH 071/136] ALSA: usb-audio: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Also, since the SG-buffer-specific PCM ops becomes identical with the normal PCM ops, unify them again to the single ops, too. Link: https://lore.kernel.org/r/20191105151856.10785-9-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 50 ++++++++----------------------------------------- 1 file changed, 8 insertions(+), 42 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ff5ab24f3bd1..9c8930bb00c8 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -785,12 +785,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - if (snd_usb_use_vmalloc) - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - else - ret = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); if (ret < 0) goto stop_pipeline; @@ -857,10 +853,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) snd_usb_unlock_shutdown(subs->stream->chip); } - if (snd_usb_use_vmalloc) - return snd_pcm_lib_free_vmalloc_buffer(substream); - else - return snd_pcm_lib_free_pages(substream); + return snd_pcm_lib_free_pages(substream); } /* @@ -1781,7 +1774,6 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_substream_playback_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops snd_usb_capture_ops = { @@ -1793,43 +1785,14 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_substream_capture_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, -}; - -static const struct snd_pcm_ops snd_usb_playback_dev_ops = { - .open = snd_usb_pcm_open, - .close = snd_usb_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_usb_hw_params, - .hw_free = snd_usb_hw_free, - .prepare = snd_usb_pcm_prepare, - .trigger = snd_usb_substream_playback_trigger, - .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, -}; - -static const struct snd_pcm_ops snd_usb_capture_dev_ops = { - .open = snd_usb_pcm_open, - .close = snd_usb_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_usb_hw_params, - .hw_free = snd_usb_hw_free, - .prepare = snd_usb_pcm_prepare, - .trigger = snd_usb_substream_capture_trigger, - .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) { const struct snd_pcm_ops *ops; - if (snd_usb_use_vmalloc) - ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? &snd_usb_playback_ops : &snd_usb_capture_ops; - else - ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_usb_playback_dev_ops : &snd_usb_capture_dev_ops; snd_pcm_set_ops(pcm, stream, ops); } @@ -1839,7 +1802,10 @@ void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; struct device *dev = subs->dev->bus->controller; - if (!snd_usb_use_vmalloc) + if (snd_usb_use_vmalloc) + snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); + else snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG, dev, 64*1024, 512*1024); } From b29e5ef15115cd77cdd250b7ec4fdb4cb8d40ddd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:41 +0100 Subject: [PATCH 072/136] ALSA: aloop: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-10-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 9ccdad89c288..54f8b17476a1 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -576,8 +576,7 @@ static void loopback_runtime_free(struct snd_pcm_runtime *runtime) static int loopback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(params)); + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } static int loopback_hw_free(struct snd_pcm_substream *substream) @@ -589,7 +588,7 @@ static int loopback_hw_free(struct snd_pcm_substream *substream) mutex_lock(&dpcm->loopback->cable_lock); cable->valid &= ~(1 << substream->stream); mutex_unlock(&dpcm->loopback->cable_lock); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static unsigned int get_cable_index(struct snd_pcm_substream *substream) @@ -765,7 +764,6 @@ static const struct snd_pcm_ops loopback_pcm_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static int loopback_pcm_new(struct loopback *loopback, @@ -780,6 +778,8 @@ static int loopback_pcm_new(struct loopback *loopback, return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_pcm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_pcm_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); pcm->private_data = loopback; pcm->info_flags = 0; From a3f4f2d5d02095d098ff03c4d0bdbae2e53a5bba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:42 +0100 Subject: [PATCH 073/136] ALSA: firewire: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-11-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_pcm.c | 9 ++++----- sound/firewire/dice/dice-pcm.c | 11 ++++++----- sound/firewire/digi00x/digi00x-pcm.c | 9 ++++----- sound/firewire/fireface/ff-pcm.c | 9 ++++----- sound/firewire/fireworks/fireworks_pcm.c | 9 ++++----- sound/firewire/isight.c | 8 ++++---- sound/firewire/motu/motu-pcm.c | 9 ++++----- sound/firewire/oxfw/oxfw-pcm.c | 14 ++++++-------- sound/firewire/tascam/tascam-pcm.c | 9 ++++----- 9 files changed, 40 insertions(+), 47 deletions(-) diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 1b100159f4c5..d4edd06d32cf 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -214,8 +214,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_bebob *bebob = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -248,7 +247,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&bebob->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int @@ -354,7 +353,6 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) .trigger = pcm_capture_trigger, .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -366,7 +364,6 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) .trigger = pcm_playback_trigger, .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -380,6 +377,8 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob) "%s PCM", bebob->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); end: return err; } diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index f1848fb39bd0..be79d659eedf 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -266,8 +266,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_dice *dice = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -305,7 +304,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&dice->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int capture_prepare(struct snd_pcm_substream *substream) @@ -417,7 +416,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .trigger = capture_trigger, .pointer = capture_pointer, .ack = capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -429,7 +427,6 @@ int snd_dice_create_pcm(struct snd_dice *dice) .trigger = playback_trigger, .pointer = playback_pointer, .ack = playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; unsigned int capture, playback; @@ -459,6 +456,10 @@ int snd_dice_create_pcm(struct snd_dice *dice) if (playback > 0) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); } return 0; diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 8befc5d2ef22..57cbce4fd836 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -190,8 +190,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_dg00x *dg00x = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -224,7 +223,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&dg00x->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int pcm_capture_prepare(struct snd_pcm_substream *substream) @@ -337,7 +336,6 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) .trigger = pcm_capture_trigger, .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -349,7 +347,6 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) .trigger = pcm_playback_trigger, .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -363,6 +360,8 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x) "%s PCM", dg00x->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index c29f87a65c0f..4e3bd9a2bec0 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -230,8 +230,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_ff *ff = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -264,7 +263,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&ff->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int pcm_capture_prepare(struct snd_pcm_substream *substream) @@ -377,7 +376,6 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff) .trigger = pcm_capture_trigger, .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops pcm_playback_ops = { .open = pcm_open, @@ -389,7 +387,6 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff) .trigger = pcm_playback_trigger, .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -403,6 +400,8 @@ int snd_ff_create_pcm_devices(struct snd_ff *ff) "%s PCM", ff->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 64c1bcf28dfa..e69896d748df 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -250,8 +250,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_efw *efw = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -284,7 +283,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&efw->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int pcm_capture_prepare(struct snd_pcm_substream *substream) @@ -384,7 +383,6 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw) .trigger = pcm_capture_trigger, .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -396,7 +394,6 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw) .trigger = pcm_playback_trigger, .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -409,6 +406,8 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw) snprintf(pcm->name, sizeof(pcm->name), "%s PCM", efw->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); end: return err; } diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index a16beda7c530..d9f1b962bfef 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -288,8 +288,7 @@ static int isight_hw_params(struct snd_pcm_substream *substream, struct isight *isight = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -337,7 +336,7 @@ static int isight_hw_free(struct snd_pcm_substream *substream) isight_stop_streaming(isight); mutex_unlock(&isight->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int isight_start_streaming(struct isight *isight) @@ -453,7 +452,6 @@ static int isight_create_pcm(struct isight *isight) .prepare = isight_prepare, .trigger = isight_trigger, .pointer = isight_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -465,6 +463,8 @@ static int isight_create_pcm(struct isight *isight) strcpy(pcm->name, "iSight"); isight->pcm = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; isight->pcm->ops = &ops; + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 490408b49157..349b4d09e84f 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -218,8 +218,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_motu *motu = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -252,7 +251,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&motu->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int capture_prepare(struct snd_pcm_substream *substream) @@ -356,7 +355,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .trigger = capture_trigger, .pointer = capture_pointer, .ack = capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -368,7 +366,6 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) .trigger = playback_trigger, .pointer = playback_pointer, .ack = playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -381,6 +378,8 @@ int snd_motu_create_pcm_devices(struct snd_motu *motu) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 74bd1811cec2..9124603edabe 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -239,8 +239,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_oxfw *oxfw = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -267,8 +266,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_oxfw *oxfw = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -303,7 +301,7 @@ static int pcm_capture_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&oxfw->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int pcm_playback_hw_free(struct snd_pcm_substream *substream) { @@ -318,7 +316,7 @@ static int pcm_playback_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&oxfw->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int pcm_capture_prepare(struct snd_pcm_substream *substream) @@ -428,7 +426,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .trigger = pcm_capture_trigger, .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -440,7 +437,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) .trigger = pcm_playback_trigger, .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; unsigned int cap = 0; @@ -458,6 +454,8 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); if (cap > 0) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index cd45f20ba515..8e9b444c8bff 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -119,8 +119,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_tscm *tscm = substream->private_data; int err; - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -153,7 +152,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) mutex_unlock(&tscm->mutex); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int pcm_capture_prepare(struct snd_pcm_substream *substream) @@ -266,7 +265,6 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) .trigger = pcm_capture_trigger, .pointer = pcm_capture_pointer, .ack = pcm_capture_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_ops = { .open = pcm_open, @@ -278,7 +276,6 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) .trigger = pcm_playback_trigger, .pointer = pcm_playback_pointer, .ack = pcm_playback_ack, - .page = snd_pcm_lib_get_vmalloc_page, }; struct snd_pcm *pcm; int err; @@ -292,6 +289,8 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm) "%s PCM", tscm->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } From 94ce90c562efd4222c1d58c52bb7dc8e2ba5827d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:43 +0100 Subject: [PATCH 074/136] ALSA: 6fire: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-12-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 4ded944e5d5d..cdc5dd7fbe16 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -449,13 +449,13 @@ static int usb6fire_pcm_close(struct snd_pcm_substream *alsa_sub) static int usb6fire_pcm_hw_params(struct snd_pcm_substream *alsa_sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_buffer(alsa_sub, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); } static int usb6fire_pcm_hw_free(struct snd_pcm_substream *alsa_sub) { - return snd_pcm_lib_free_vmalloc_buffer(alsa_sub); + return snd_pcm_lib_free_pages(alsa_sub); } static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub) @@ -560,7 +560,6 @@ static const struct snd_pcm_ops pcm_ops = { .prepare = usb6fire_pcm_prepare, .trigger = usb6fire_pcm_trigger, .pointer = usb6fire_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static void usb6fire_pcm_init_urb(struct pcm_urb *urb, @@ -659,6 +658,8 @@ int usb6fire_pcm_init(struct sfire_chip *chip) strcpy(pcm->name, "DMX 6Fire USB"); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); rt->instance = pcm; From c7b03852e74dc294d0f09c2c6f3d5e28615ce689 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:44 +0100 Subject: [PATCH 075/136] ALSA: caiaq: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-13-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 444bb637ce13..970eb0865ba3 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -170,15 +170,14 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_buffer(sub, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); } static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) { struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(sub); deactivate_substream(cdev, sub); - return snd_pcm_lib_free_vmalloc_buffer(sub); + return snd_pcm_lib_free_pages(sub); } /* this should probably go upstream */ @@ -334,7 +333,6 @@ static const struct snd_pcm_ops snd_usb_caiaq_ops = { .prepare = snd_usb_caiaq_pcm_prepare, .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, @@ -843,6 +841,8 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) &snd_usb_caiaq_ops); snd_pcm_set_ops(cdev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); + snd_pcm_lib_preallocate_pages_for_all(cdev->pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); cdev->data_cb_info = kmalloc_array(N_URBS, sizeof(struct snd_usb_caiaq_cb_info), From d7867ee7d264d4760214de2a1cb10c694d935dab Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:45 +0100 Subject: [PATCH 076/136] ALSA: hiface: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-14-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/hiface/pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index c406497c5919..e0de71917274 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -418,13 +418,13 @@ static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_buffer(alsa_sub, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); } static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) { - return snd_pcm_lib_free_vmalloc_buffer(alsa_sub); + return snd_pcm_lib_free_pages(alsa_sub); } static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) @@ -518,7 +518,6 @@ static const struct snd_pcm_ops pcm_ops = { .prepare = hiface_pcm_prepare, .trigger = hiface_pcm_trigger, .pointer = hiface_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static int hiface_pcm_init_urb(struct pcm_urb *urb, @@ -614,6 +613,8 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) strlcpy(pcm->name, "USB-SPDIF Audio", sizeof(pcm->name)); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); rt->instance = pcm; From 95ef15c65ae0d36569ac374e83498b6bf08d861f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:46 +0100 Subject: [PATCH 077/136] ALSA: ua101: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-15-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/misc/ua101.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 307b72d5fffa..566a4a31528a 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -733,8 +733,8 @@ static int capture_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); } static int playback_pcm_hw_params(struct snd_pcm_substream *substream, @@ -751,13 +751,13 @@ static int playback_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); } static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) { - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int capture_pcm_prepare(struct snd_pcm_substream *substream) @@ -889,7 +889,6 @@ static const struct snd_pcm_ops capture_pcm_ops = { .prepare = capture_pcm_prepare, .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops playback_pcm_ops = { @@ -901,7 +900,6 @@ static const struct snd_pcm_ops playback_pcm_ops = { .prepare = playback_pcm_prepare, .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct uac_format_type_i_discrete_descriptor * @@ -1296,6 +1294,8 @@ static int ua101_probe(struct usb_interface *interface, strcpy(ua->pcm->name, name); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); + snd_pcm_lib_preallocate_pages_for_all(ua->pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); err = snd_usbmidi_create(card, ua->intf[INTF_MIDI], &ua->midi_list, &midi_quirk); From 748a76b215a36e9bb88a5bd8b56c5088371b2f19 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:47 +0100 Subject: [PATCH 078/136] ALSA: vx: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Since the driver requires the DMA32 allocation, it passes the specially encoded device to snd_pcm_lib_preallocate_pages(). Link: https://lore.kernel.org/r/20191105151856.10785-16-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 4705c50fbf4f..f17e0a76c73c 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -778,8 +778,7 @@ static snd_pcm_uframes_t vx_pcm_playback_pointer(struct snd_pcm_substream *subs) static int vx_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_32_buffer - (subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw_params)); } /* @@ -787,7 +786,7 @@ static int vx_pcm_hw_params(struct snd_pcm_substream *subs, */ static int vx_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_lib_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_pages(subs); } /* @@ -867,7 +866,6 @@ static const struct snd_pcm_ops vx_pcm_playback_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; @@ -1088,7 +1086,6 @@ static const struct snd_pcm_ops vx_pcm_capture_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; @@ -1233,6 +1230,9 @@ int snd_vx_pcm_new(struct vx_core *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &vx_pcm_playback_ops); if (ins) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &vx_pcm_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + snd_dma_continuous_data(GFP_KERNEL | GFP_DMA32), + 0, 0); pcm->private_data = chip; pcm->private_free = snd_vx_pcm_free; From d5dec270c00c155db64c3d2151b94002dfb2c6c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:48 +0100 Subject: [PATCH 079/136] ALSA: pdaudiocf: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Since the driver requires the DMA32 allocation, it passes the specially encoded device to snd_pcm_lib_preallocate_pages(). Link: https://lore.kernel.org/r/20191105151856.10785-17-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index c21fec60cd98..067b1c3a3e02 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -89,8 +89,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_32_buffer - (subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw_params)); } /* @@ -98,7 +97,7 @@ static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, */ static int pdacf_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_lib_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_pages(subs); } /* @@ -262,7 +261,6 @@ static const struct snd_pcm_ops pdacf_pcm_capture_ops = { .prepare = pdacf_pcm_prepare, .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; @@ -279,6 +277,9 @@ int snd_pdacf_pcm_new(struct snd_pdacf *chip) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pdacf_pcm_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + snd_dma_continuous_data(GFP_KERNEL | GFP_DMA32), + 0, 0); pcm->private_data = chip; pcm->info_flags = 0; From 30f2ce712d29325b0150243a9a2dd215c2ff60c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:49 +0100 Subject: [PATCH 080/136] ALSA: mips: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-18-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index fadc1194b136..9d20ce6118a0 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -582,14 +582,13 @@ static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } /* hw_free callback */ static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) { - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } /* prepare callback */ @@ -670,7 +669,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -682,7 +680,6 @@ static const struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -694,7 +691,6 @@ static const struct snd_pcm_ops snd_sgio2audio_capture_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; /* @@ -720,6 +716,8 @@ static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) &snd_sgio2audio_playback1_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sgio2audio_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); /* create second pcm device with one outputs and no input */ err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm); @@ -732,6 +730,8 @@ static int snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) /* set operators */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sgio2audio_playback2_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; } From 16ccca11088c1bdd9311a2c630b453541305c48a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:50 +0100 Subject: [PATCH 081/136] ALSA: pci: Drop superfluous snd_pcm_sgbuf_ops_page snd_pcm_sgbuf_ops_page is no longer needed to be set explicitly to PCM page ops since the recent change in the PCM core (*). Leaving it NULL should work as long as the preallocation has been done properly. This patch drops the redundant lines. (*) 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Link: https://lore.kernel.org/r/20191105151856.10785-19-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_pcm.c | 1 - sound/pci/bt87x.c | 1 - sound/pci/ctxfi/ctpcm.c | 2 -- sound/pci/echoaudio/echoaudio.c | 4 ---- sound/pci/emu10k1/emupcm.c | 2 -- sound/pci/hda/hda_controller.c | 1 - sound/pci/lola/lola_pcm.c | 1 - sound/pci/riptide/riptide.c | 2 -- sound/pci/rme9652/hdspm.c | 1 - sound/pci/trident/trident_main.c | 2 -- sound/pci/via82xx.c | 5 ----- sound/pci/via82xx_modem.c | 2 -- 12 files changed, 24 deletions(-) diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 39ea9ef00f47..b835042893d9 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -436,7 +436,6 @@ static const struct snd_pcm_ops snd_vortex_playback_ops = { .prepare = snd_vortex_pcm_prepare, .trigger = snd_vortex_pcm_trigger, .pointer = snd_vortex_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 66a5a24e7558..d4b6da79ef49 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -545,7 +545,6 @@ static const struct snd_pcm_ops snd_bt87x_pcm_ops = { .prepare = snd_bt87x_prepare, .trigger = snd_bt87x_trigger, .pointer = snd_bt87x_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static int snd_bt87x_capture_volume_info(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 89923399e646..c341f509e4fc 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -379,7 +379,6 @@ static const struct snd_pcm_ops ct_pcm_playback_ops = { .prepare = ct_pcm_playback_prepare, .trigger = ct_pcm_playback_trigger, .pointer = ct_pcm_playback_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* PCM operators for capture */ @@ -392,7 +391,6 @@ static const struct snd_pcm_ops ct_pcm_capture_ops = { .prepare = ct_pcm_capture_prepare, .trigger = ct_pcm_capture_trigger, .pointer = ct_pcm_capture_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static const struct snd_pcm_chmap_elem surround_map[] = { diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index ca9125726be2..40877e478bb6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -824,7 +824,6 @@ static const struct snd_pcm_ops analog_playback_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static const struct snd_pcm_ops analog_capture_ops = { .open = pcm_analog_in_open, @@ -835,7 +834,6 @@ static const struct snd_pcm_ops analog_capture_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; #ifdef ECHOCARD_HAS_DIGITAL_IO #ifndef ECHOCARD_HAS_VMIXER @@ -848,7 +846,6 @@ static const struct snd_pcm_ops digital_playback_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; #endif /* !ECHOCARD_HAS_VMIXER */ static const struct snd_pcm_ops digital_capture_ops = { @@ -860,7 +857,6 @@ static const struct snd_pcm_ops digital_capture_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; #endif /* ECHOCARD_HAS_DIGITAL_IO */ diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 6530a55fb878..46d8768adc01 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1366,7 +1366,6 @@ static const struct snd_pcm_ops snd_emu10k1_playback_ops = { .prepare = snd_emu10k1_playback_prepare, .trigger = snd_emu10k1_playback_trigger, .pointer = snd_emu10k1_playback_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static const struct snd_pcm_ops snd_emu10k1_capture_ops = { @@ -1390,7 +1389,6 @@ static const struct snd_pcm_ops snd_emu10k1_efx_playback_ops = { .prepare = snd_emu10k1_efx_playback_prepare, .trigger = snd_emu10k1_efx_playback_trigger, .pointer = snd_emu10k1_efx_playback_pointer, - .page = snd_pcm_sgbuf_ops_page, }; int snd_emu10k1_pcm(struct snd_emu10k1 *emu, int device) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6387c7e90918..2f3b7a35f2d9 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -701,7 +701,6 @@ static const struct snd_pcm_ops azx_pcm_ops = { .pointer = azx_pcm_pointer, .get_time_info = azx_get_time_info, .mmap = azx_pcm_mmap, - .page = snd_pcm_sgbuf_ops_page, }; static void azx_pcm_free(struct snd_pcm *pcm) diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 151f7cf5ce0e..f032d7ceb6e5 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -582,7 +582,6 @@ static const struct snd_pcm_ops lola_pcm_ops = { .prepare = lola_pcm_prepare, .trigger = lola_pcm_trigger, .pointer = lola_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; int lola_create_pcm(struct lola *chip) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 58771ae0ed63..ac10d6f229d8 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1661,7 +1661,6 @@ static const struct snd_pcm_ops snd_riptide_playback_ops = { .hw_params = snd_riptide_hw_params, .hw_free = snd_riptide_hw_free, .prepare = snd_riptide_prepare, - .page = snd_pcm_sgbuf_ops_page, .trigger = snd_riptide_trigger, .pointer = snd_riptide_pointer, }; @@ -1672,7 +1671,6 @@ static const struct snd_pcm_ops snd_riptide_capture_ops = { .hw_params = snd_riptide_hw_params, .hw_free = snd_riptide_hw_free, .prepare = snd_riptide_prepare, - .page = snd_pcm_sgbuf_ops_page, .trigger = snd_riptide_trigger, .pointer = snd_riptide_pointer, }; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 81a6f4b2bd3c..b571d9d017bb 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6368,7 +6368,6 @@ static const struct snd_pcm_ops snd_hdspm_ops = { .prepare = snd_hdspm_prepare, .trigger = snd_hdspm_trigger, .pointer = snd_hdspm_hw_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static int snd_hdspm_create_hwdep(struct snd_card *card, diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 1a6f6202fd16..8ac6f541b008 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -2076,7 +2076,6 @@ static const struct snd_pcm_ops snd_trident_nx_playback_ops = { .prepare = snd_trident_playback_prepare, .trigger = snd_trident_trigger, .pointer = snd_trident_playback_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static const struct snd_pcm_ops snd_trident_capture_ops = { @@ -2121,7 +2120,6 @@ static const struct snd_pcm_ops snd_trident_nx_foldback_ops = { .prepare = snd_trident_foldback_prepare, .trigger = snd_trident_trigger, .pointer = snd_trident_playback_pointer, - .page = snd_pcm_sgbuf_ops_page, }; static const struct snd_pcm_ops snd_trident_spdif_ops = { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 38601d0dfb73..902409dbc4c8 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1363,7 +1363,6 @@ static const struct snd_pcm_ops snd_via686_playback_ops = { .prepare = snd_via686_playback_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* via686 capture callbacks */ @@ -1376,7 +1375,6 @@ static const struct snd_pcm_ops snd_via686_capture_ops = { .prepare = snd_via686_capture_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* via823x DSX playback callbacks */ @@ -1389,7 +1387,6 @@ static const struct snd_pcm_ops snd_via8233_playback_ops = { .prepare = snd_via8233_playback_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via8233_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* via823x multi-channel playback callbacks */ @@ -1402,7 +1399,6 @@ static const struct snd_pcm_ops snd_via8233_multi_ops = { .prepare = snd_via8233_multi_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via8233_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* via823x capture callbacks */ @@ -1415,7 +1411,6 @@ static const struct snd_pcm_ops snd_via8233_capture_ops = { .prepare = snd_via8233_capture_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via8233_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index bfb5e1b89d5f..2c8370cf03b3 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -801,7 +801,6 @@ static const struct snd_pcm_ops snd_via686_playback_ops = { .prepare = snd_via82xx_pcm_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; /* via686 capture callbacks */ @@ -814,7 +813,6 @@ static const struct snd_pcm_ops snd_via686_capture_ops = { .prepare = snd_via82xx_pcm_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, }; From 3ca5fc0664ec47fae86bc07749ceb185f7b175f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:51 +0100 Subject: [PATCH 082/136] ALSA: aoa: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Link: https://lore.kernel.org/r/20191105151856.10785-20-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 7f0754dd3d7d..a94e4023fadf 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -1028,7 +1028,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, /* well, we really should support scatter/gather DMA */ snd_pcm_lib_preallocate_pages_for_all( dev->pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(macio_get_pci_dev(i2sdev->macio)), + &macio_get_pci_dev(i2sdev->macio)->dev, 64 * 1024, 64 * 1024); return 0; From 7564d3b6334c5dd619827a85dd32bf7c21df6667 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:52 +0100 Subject: [PATCH 083/136] ALSA: echoaudio: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Also slightly refactor the code (drop the return value check from the preallocation) as it never returns an error. Link: https://lore.kernel.org/r/20191105151856.10785-21-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 40877e478bb6..1465813bf7c6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -324,7 +324,7 @@ static int pcm_open(struct snd_pcm_substream *substream, /* Finally allocate a page for the scatter-gather list */ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, PAGE_SIZE, &pipe->sgpage)) < 0) { dev_err(chip->card->dev, "s-g list allocation failed\n"); return err; @@ -865,7 +865,7 @@ static const struct snd_pcm_ops digital_capture_ops = { /* Preallocate memory only for the first substream because it's the most * used one */ -static int snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev) +static void snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev) { struct snd_pcm_substream *ss; int stream; @@ -876,8 +876,6 @@ static int snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev) dev, ss->number ? 0 : 128<<10, 256<<10); - - return 0; } @@ -904,8 +902,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) strcpy(pcm->name, chip->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); - if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) - return err; + snd_echo_preallocate_pages(pcm, &chip->pci->dev); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital inputs, no outputs */ @@ -916,8 +913,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) chip->digital_pcm = pcm; strcpy(pcm->name, chip->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); - if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) - return err; + snd_echo_preallocate_pages(pcm, &chip->pci->dev); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #else /* ECHOCARD_HAS_VMIXER */ @@ -937,8 +933,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) strcpy(pcm->name, chip->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); - if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) - return err; + snd_echo_preallocate_pages(pcm, &chip->pci->dev); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital i/o */ @@ -951,8 +946,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) strcpy(pcm->name, chip->card->shortname); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &digital_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); - if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) - return err; + snd_echo_preallocate_pages(pcm, &chip->pci->dev); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #endif /* ECHOCARD_HAS_VMIXER */ @@ -1954,7 +1948,7 @@ static int snd_echo_create(struct snd_card *card, /* Create the DSP comm page - this is the area of memory used for most of the communication with the DSP, which accesses it via bus mastering */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, sizeof(struct comm_page), &chip->commpage_dma_buf) < 0) { dev_err(chip->card->dev, "cannot allocate the comm page\n"); From af7153a29d5f0bed1f62b5347dc6cce291d40d69 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:53 +0100 Subject: [PATCH 084/136] ALSA: es1968: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Along with it, the unneeded assignment before snd_dma_alloc_pages*() call is dropped. Link: https://lore.kernel.org/r/20191105151856.10785-22-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 974142535a25..7017ca9dea4a 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1422,10 +1422,8 @@ snd_es1968_init_dmabuf(struct es1968 *chip) int err; struct esm_memory *chunk; - chip->dma.dev.type = SNDRV_DMA_TYPE_DEV; - chip->dma.dev.dev = snd_dma_pci_data(chip->pci); err = snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, chip->total_bufsize, &chip->dma); if (err < 0 || ! chip->dma.area) { dev_err(chip->card->dev, From 1a810436a2eff372aa1a934fe4d1bef50bd9dac8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:54 +0100 Subject: [PATCH 085/136] ALSA: rme: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Along with it, drop the unnecessary assignment before the snd_dma_alloc_pages() call and simplify by returning the error code directly. Link: https://lore.kernel.org/r/20191105151856.10785-23-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 7 +------ sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 7 +------ 3 files changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 5cbdc9be9c7e..cd20af465d8e 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -569,12 +569,7 @@ static char channel_map_H9632_qs[HDSP_MAX_CHANNELS] = { static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer *dmab, size_t size) { - dmab->dev.type = SNDRV_DMA_TYPE_DEV; - dmab->dev.dev = snd_dma_pci_data(pci); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - size, dmab) < 0) - return -ENOMEM; - return 0; + return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, size, dmab); } static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b571d9d017bb..75c06a7cc779 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6406,7 +6406,7 @@ static int snd_hdspm_preallocate_memory(struct hdspm *hdspm) wanted = HDSPM_DMA_AREA_BYTES; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(hdspm->pci), + &hdspm->pci->dev, wanted, wanted); dev_dbg(hdspm->card->dev, " Preallocated %zd Bytes\n", wanted); return 0; diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 4c851f8dcaf8..ef5c2f8e17c7 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -279,12 +279,7 @@ static char channel_map_9636_ds[26] = { static int snd_hammerfall_get_buffer(struct pci_dev *pci, struct snd_dma_buffer *dmab, size_t size) { - dmab->dev.type = SNDRV_DMA_TYPE_DEV; - dmab->dev.dev = snd_dma_pci_data(pci); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - size, dmab) < 0) - return -ENOMEM; - return 0; + return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, size, dmab); } static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci) From 6974f8ad44946701779209cb03cd8c6b598c3342 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:55 +0100 Subject: [PATCH 086/136] ALSA: pci: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Link: https://lore.kernel.org/r/20191105151856.10785-24-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/ad1889.c | 6 +++--- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 3 ++- sound/pci/als4000.c | 3 ++- sound/pci/asihpi/asihpi.c | 4 ++-- sound/pci/atiixp.c | 6 +++--- sound/pci/atiixp_modem.c | 4 ++-- sound/pci/au88x0/au88x0_pcm.c | 2 +- sound/pci/aw2/aw2-alsa.c | 6 +++--- sound/pci/azt3328.c | 8 ++++---- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 6 +++--- sound/pci/cmipci.c | 6 +++--- sound/pci/cs4281.c | 3 ++- sound/pci/cs46xx/cs46xx_lib.c | 16 ++++++++++------ sound/pci/cs5535audio/cs5535audio_pcm.c | 6 +++--- sound/pci/ctxfi/ctpcm.c | 3 ++- sound/pci/ctxfi/ctvmem.c | 2 +- sound/pci/emu10k1/emu10k1.c | 5 +++-- sound/pci/emu10k1/emu10k1x.c | 6 +++--- sound/pci/emu10k1/emufx.c | 2 +- sound/pci/emu10k1/emupcm.c | 10 +++++----- sound/pci/emu10k1/memory.c | 4 ++-- sound/pci/emu10k1/p16v.c | 4 ++-- sound/pci/ens1370.c | 8 +++++--- sound/pci/es1938.c | 3 ++- sound/pci/fm801.c | 2 +- sound/pci/ice1712/ice1712.c | 9 ++++++--- sound/pci/ice1712/ice1724.c | 6 +++--- sound/pci/intel8x0.c | 4 ++-- sound/pci/intel8x0m.c | 4 ++-- sound/pci/korg1212/korg1212.c | 8 ++++---- sound/pci/lola/lola.c | 2 +- sound/pci/lola/lola_pcm.c | 4 ++-- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 3 ++- sound/pci/mixart/mixart.c | 7 ++++--- sound/pci/oxygen/oxygen_pcm.c | 10 +++++----- sound/pci/pcxhr/pcxhr.c | 4 ++-- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/sis7019.c | 3 ++- sound/pci/sonicvibes.c | 3 ++- sound/pci/trident/trident_main.c | 22 ++++++++++++++-------- sound/pci/via82xx.c | 12 ++++++------ sound/pci/via82xx_modem.c | 4 ++-- sound/pci/ymfpci/ymfpci_main.c | 16 ++++++++++------ 46 files changed, 145 insertions(+), 116 deletions(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 4b2451287e2c..5b6452df8bbd 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -633,9 +633,9 @@ snd_ad1889_pcm_init(struct snd_ad1889 *chip, int device) chip->csubs = NULL; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - BUFFER_BYTES_MAX / 2, - BUFFER_BYTES_MAX); + &chip->pci->dev, + BUFFER_BYTES_MAX / 2, + BUFFER_BYTES_MAX); return 0; } diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 6e28e381c21a..ae29df085ae1 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -1672,7 +1672,7 @@ static int snd_ali_pcm(struct snd_ali *codec, int device, desc->capture_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(codec->pci), + &codec->pci->dev, 64*1024, 128*1024); pcm->info_flags = 0; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 530799c8d3ce..cfbb8cacaaac 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -592,7 +592,8 @@ static int snd_als300_new_pcm(struct snd_als300 *chip) /* pre-allocation of buffers */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 64*1024); + &chip->pci->dev, + 64*1024, 64*1024); return 0; } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index b06c3dbb525d..d6f5487afe52 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -693,7 +693,8 @@ static int snd_als4000_pcm(struct snd_sb *chip, int device) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_als4000_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_als4000_capture_ops); - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pci->dev, 64*1024, 64*1024); chip->pcm = pcm; diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 2a21a3d99719..147005fdd3ea 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1325,8 +1325,8 @@ static int snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi, int device) /*? do we want to emulate MMAP for non-BBM cards? Jack doesn't work with ALSAs MMAP emulation - WHY NOT? */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(asihpi->pci), - 64*1024, BUFFER_BYTES_MAX); + &asihpi->pci->dev, + 64*1024, BUFFER_BYTES_MAX); return 0; } diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index c953bd73a48c..1e1ededf8eb2 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -353,7 +353,7 @@ static int atiixp_build_dma_packets(struct atiixp *chip, struct atiixp_dma *dma, if (dma->desc_buf.area == NULL) { if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, ATI_DESC_LIST_SIZE, &dma->desc_buf) < 0) return -ENOMEM; @@ -1284,7 +1284,7 @@ static int snd_atiixp_pcm_new(struct atiixp *chip) chip->pcmdevs[ATI_PCMDEV_ANALOG] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, 128*1024); err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -1317,7 +1317,7 @@ static int snd_atiixp_pcm_new(struct atiixp *chip) chip->pcmdevs[ATI_PCMDEV_DIGITAL] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, 128*1024); /* pre-select AC97 SPDIF slots 10/11 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 95d209f96581..6f088c1949f3 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -321,7 +321,7 @@ static int atiixp_build_dma_packets(struct atiixp_modem *chip, return -ENOMEM; if (dma->desc_buf.area == NULL) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, ATI_DESC_LIST_SIZE, &dma->desc_buf) < 0) return -ENOMEM; dma->period_bytes = dma->periods = 0; /* clear */ @@ -995,7 +995,7 @@ static int snd_atiixp_pcm_new(struct atiixp_modem *chip) chip->pcmdevs[ATI_PCMDEV_ANALOG] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, 128*1024); return 0; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index b835042893d9..1012e8da29e7 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -637,7 +637,7 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) /* pre-allocation of Scatter-Gather buffers */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci_dev), + &chip->pci_dev, 0x10000, 0x10000); switch (VORTEX_PCM_TYPE(pcm)) { diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index e413414181df..1cbfae856a2a 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -613,7 +613,7 @@ static int snd_aw2_new_pcm(struct aw2 *chip) /* Preallocate continuous pages. */ snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64 * 1024, 64 * 1024); err = snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0, @@ -645,7 +645,7 @@ static int snd_aw2_new_pcm(struct aw2 *chip) /* Preallocate continuous pages. */ snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64 * 1024, 64 * 1024); err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1, @@ -678,7 +678,7 @@ static int snd_aw2_new_pcm(struct aw2 *chip) /* Preallocate continuous pages. */ snd_pcm_lib_preallocate_pages_for_all(pcm_capture, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64 * 1024, 64 * 1024); /* Create control */ diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index f92c9cbb955a..f475370faaaa 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2135,8 +2135,8 @@ snd_azf3328_pcm(struct snd_azf3328 *chip) chip->pcm[AZF_CODEC_CAPTURE] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 64*1024, 64*1024); + &chip->pci->dev, + 64*1024, 64*1024); err = snd_pcm_new(chip->card, "AZF3328 I2S OUT", AZF_PCMDEV_I2S_OUT, 1, 0, &pcm); @@ -2151,8 +2151,8 @@ snd_azf3328_pcm(struct snd_azf3328 *chip) chip->pcm[AZF_CODEC_I2S_OUT] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 64*1024, 64*1024); + &chip->pci->dev, + 64*1024, 64*1024); return 0; } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index d4b6da79ef49..6bf5ac3600c5 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -217,7 +217,7 @@ static int snd_bt87x_create_risc(struct snd_bt87x *chip, struct snd_pcm_substrea __le32 *risc; if (chip->dma_risc.area == NULL) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, PAGE_ALIGN(MAX_RISC_SIZE), &chip->dma_risc) < 0) return -ENOMEM; } @@ -700,7 +700,7 @@ static int snd_bt87x_pcm(struct snd_bt87x *chip, int device, char *name) strcpy(pcm->name, name); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_bt87x_pcm_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 128 * 1024, ALIGN(255 * 4092, 1024)); return 0; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 478412e0aa3c..abc2440dc2d9 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1389,7 +1389,7 @@ static int snd_ca0106_pcm(struct snd_ca0106 *emu, int device) substream; substream = substream->next) { snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); } @@ -1397,7 +1397,7 @@ static int snd_ca0106_pcm(struct snd_ca0106 *emu, int device) substream; substream = substream->next) { snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); } @@ -1692,7 +1692,7 @@ static int snd_ca0106_create(int dev, struct snd_card *card, chip->irq = pci->irq; /* This stores the periods table. */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, 1024, &chip->buffer) < 0) { snd_ca0106_free(chip); return -ENOMEM; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index df720881eb99..dd9d62e2b633 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1902,7 +1902,7 @@ static int snd_cmipci_pcm_new(struct cmipci *cm, int device) cm->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(cm->pci), 64*1024, 128*1024); + &cm->pci->dev, 64*1024, 128*1024); return 0; } @@ -1924,7 +1924,7 @@ static int snd_cmipci_pcm2_new(struct cmipci *cm, int device) cm->pcm2 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(cm->pci), 64*1024, 128*1024); + &cm->pci->dev, 64*1024, 128*1024); return 0; } @@ -1947,7 +1947,7 @@ static int snd_cmipci_pcm_spdif_new(struct cmipci *cm, int device) cm->pcm_spdif = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(cm->pci), 64*1024, 128*1024); + &cm->pci->dev, 64*1024, 128*1024); err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, snd_pcm_alt_chmaps, cm->max_channels, 0, diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 04c712647853..058c1414b777 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -975,7 +975,8 @@ static int snd_cs4281_pcm(struct cs4281 *chip, int device) chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 512*1024); + &chip->pci->dev, + 64*1024, 512*1024); return 0; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 5b888b795f7e..102a62965ac1 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1494,7 +1494,7 @@ static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,in cpcm = kzalloc(sizeof(*cpcm), GFP_KERNEL); if (cpcm == NULL) return -ENOMEM; - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, PAGE_SIZE, &cpcm->hw_buf) < 0) { kfree(cpcm); return -ENOMEM; @@ -1582,7 +1582,7 @@ static int snd_cs46xx_capture_open(struct snd_pcm_substream *substream) { struct snd_cs46xx *chip = snd_pcm_substream_chip(substream); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, PAGE_SIZE, &chip->capt.hw_buf) < 0) return -ENOMEM; chip->capt.substream = substream; @@ -1784,7 +1784,8 @@ int snd_cs46xx_pcm(struct snd_cs46xx *chip, int device) chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return 0; } @@ -1809,7 +1810,8 @@ int snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device) chip->pcm_rear = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return 0; } @@ -1832,7 +1834,8 @@ int snd_cs46xx_pcm_center_lfe(struct snd_cs46xx *chip, int device) chip->pcm_center_lfe = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return 0; } @@ -1855,7 +1858,8 @@ int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device) chip->pcm_iec958 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return 0; } diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 04822bf2f987..4642e5384e83 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -117,7 +117,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, if (dma->desc_buf.area == NULL) { if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(cs5535au->pci), + &cs5535au->pci->dev, CS5535AUDIO_DESC_LIST_SIZE+1, &dma->desc_buf) < 0) return -ENOMEM; @@ -432,8 +432,8 @@ int snd_cs5535audio_pcm(struct cs5535audio *cs5535au) strcpy(pcm->name, "CS5535 Audio"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(cs5535au->pci), - 64*1024, 128*1024); + &cs5535au->pci->dev, + 64*1024, 128*1024); cs5535au->pcm = pcm; return 0; diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index c341f509e4fc..7ae5b238703c 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -450,7 +450,8 @@ int ct_alsa_pcm_create(struct ct_atc *atc, SNDRV_PCM_STREAM_CAPTURE, &ct_pcm_capture_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(atc->pci), 128*1024, 128*1024); + &atc->pci->dev, + 128*1024, 128*1024); chs = 2; switch (device) { diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 2e80b17a7104..bde28aa9e139 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -183,7 +183,7 @@ int ct_vm_create(struct ct_vm **rvm, struct pci_dev *pci) /* Allocate page table pages */ for (i = 0; i < CT_PTP_NUM; i++) { err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(pci), + &pci->dev, PAGE_SIZE, &vm->ptp[i]); if (err < 0) break; diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index f208b6e217fd..29b7720d7961 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -124,8 +124,9 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, goto error; /* This stores the periods table. */ if (emu->card_capabilities->ca0151_chip) { /* P16V */ - if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - 1024, &emu->p16v_buffer)) < 0) + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, + 1024, &emu->p16v_buffer); + if (err < 0) goto error; } diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 9cf81832259c..241b4a0631ab 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -877,7 +877,7 @@ static int snd_emu10k1x_pcm(struct emu10k1x *emu, int device) emu->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 32*1024, 32*1024); return snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, 2, @@ -936,8 +936,8 @@ static int snd_emu10k1x_create(struct snd_card *card, } chip->irq = pci->irq; - if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - 4 * 1024, &chip->dma_buffer) < 0) { + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, + 4 * 1024, &chip->dma_buffer) < 0) { snd_emu10k1x_free(chip); return -ENOMEM; } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index e053f0d58bdd..a31adecfe608 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2464,7 +2464,7 @@ int snd_emu10k1_fx8010_tram_setup(struct snd_emu10k1 *emu, u32 size) } if (size > 0) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &emu->pci->dev, size * 2, &emu->fx8010.etram_pages) < 0) return -ENOMEM; memset(emu->fx8010.etram_pages.area, 0, size * 2); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 46d8768adc01..9a8cf3c7dd67 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1412,12 +1412,12 @@ int snd_emu10k1_pcm(struct snd_emu10k1 *emu, int device) for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next) snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); return 0; @@ -1443,7 +1443,7 @@ int snd_emu10k1_pcm_multi(struct snd_emu10k1 *emu, int device) for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); return 0; @@ -1478,7 +1478,7 @@ int snd_emu10k1_pcm_mic(struct snd_emu10k1 *emu, int device) emu->pcm_mic = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); return 0; @@ -1853,7 +1853,7 @@ int snd_emu10k1_pcm_efx(struct snd_emu10k1 *emu, int device) return err; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 64*1024, 64*1024); return 0; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 135e26544275..94b8d5b08225 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -387,7 +387,7 @@ int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, } return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), size, dmab); + &emu->pci->dev, size, dmab); } /* @@ -477,7 +477,7 @@ static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, int page; dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(emu->pci); + dmab.dev.dev = &emu->pci->dev; for (page = first_page; page <= last_page; page++) { if (emu->page_ptr_table[page] == NULL) diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index eeaed555185c..ab8876855989 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -643,7 +643,7 @@ int snd_p16v_pcm(struct snd_emu10k1 *emu, int device) substream; substream = substream->next) { snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, (65536 - 64) * 8, (65536 - 64) * 8); /* @@ -656,7 +656,7 @@ int snd_p16v_pcm(struct snd_emu10k1 *emu, int device) substream; substream = substream->next) { snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), + &emu->pci->dev, 65536 - 64, 65536 - 64); /* dev_dbg(emu->card->dev, diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index b767df8181b5..0499dc863202 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1275,7 +1275,8 @@ static int snd_ensoniq_pcm(struct ensoniq *ensoniq, int device) ensoniq->pcm1 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ensoniq->pci), 64*1024, 128*1024); + &ensoniq->pci->dev, + 64*1024, 128*1024); #ifdef CHIP1370 err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -1307,7 +1308,8 @@ static int snd_ensoniq_pcm2(struct ensoniq *ensoniq, int device) ensoniq->pcm2 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ensoniq->pci), 64*1024, 128*1024); + &ensoniq->pci->dev, + 64*1024, 128*1024); #ifdef CHIP1370 err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -2095,7 +2097,7 @@ static int snd_ensoniq_create(struct snd_card *card, } ensoniq->irq = pci->irq; #ifdef CHIP1370 - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, 16, &ensoniq->dma_bug) < 0) { dev_err(card->dev, "unable to allocate space for phantom area - dma_bug\n"); snd_ensoniq_free(ensoniq); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index ecf77c8c9e59..c571c5d380ca 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1032,7 +1032,8 @@ static int snd_es1938_new_pcm(struct es1938 *chip, int device) strcpy(pcm->name, "ESS Solo-1"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 64*1024); + &chip->pci->dev, + 64*1024, 64*1024); chip->pcm = pcm; return 0; diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 3ef7d507eb9b..a7f8109acced 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -721,7 +721,7 @@ static int snd_fm801_pcm(struct fm801 *chip, int device) chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(pdev), + &pdev->dev, chip->multichannel ? 128*1024 : 64*1024, 128*1024); return snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 4b0dea7f7669..deadba40131c 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -884,7 +884,8 @@ static int snd_ice1712_pcm(struct snd_ice1712 *ice, int device) ice->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ice->pci), 64*1024, 64*1024); + &ice->pci->dev, + 64*1024, 64*1024); dev_warn(ice->card->dev, "Consumer PCM code does not work well at the moment --jk\n"); @@ -909,7 +910,8 @@ static int snd_ice1712_pcm_ds(struct snd_ice1712 *ice, int device) ice->pcm_ds = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ice->pci), 64*1024, 128*1024); + &ice->pci->dev, + 64*1024, 128*1024); return 0; } @@ -1253,7 +1255,8 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device) strcpy(pcm->name, "ICE1712 multi"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ice->pci), 256*1024, 256*1024); + &ice->pci->dev, + 256*1024, 256*1024); ice->pcm_pro = pcm; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index e62c11816683..c80a16ee6e76 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1143,7 +1143,7 @@ static int snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) strcpy(pcm->name, "ICE1724"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ice->pci), + &ice->pci->dev, 256*1024, 256*1024); ice->pcm_pro = pcm; @@ -1341,7 +1341,7 @@ static int snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) strcpy(pcm->name, name); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ice->pci), + &ice->pci->dev, 256*1024, 256*1024); ice->pcm = pcm; @@ -1455,7 +1455,7 @@ static int snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device) strcpy(pcm->name, "ICE1724 Surround PCM"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(ice->pci), + &ice->pci->dev, 256*1024, 256*1024); ice->pcm_ds = pcm; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6ff94d8ad86e..12374ba08ca2 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1488,7 +1488,7 @@ static int snd_intel8x0_pcm1(struct intel8x0 *chip, int device, chip->pcm[device] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, intel8x0_dma_type(chip), - snd_dma_pci_data(chip->pci), + &chip->pci->dev, rec->prealloc_size, rec->prealloc_max_size); if (rec->playback_ops && @@ -3047,7 +3047,7 @@ static int snd_intel8x0_create(struct snd_card *card, /* allocate buffer descriptor lists */ /* the start of each lists must be aligned to 8 bytes */ - if (snd_dma_alloc_pages(intel8x0_dma_type(chip), snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(intel8x0_dma_type(chip), &pci->dev, chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2, &chip->bdbars) < 0) { snd_intel8x0_free(chip); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 2f960fb092df..a9add5fedfcb 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -734,7 +734,7 @@ static int snd_intel8x0m_pcm1(struct intel8x0m *chip, int device, chip->pcm[device] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, rec->prealloc_size, rec->prealloc_max_size); @@ -1176,7 +1176,7 @@ static int snd_intel8x0m_create(struct snd_card *card, /* allocate buffer descriptor lists */ /* the start of each lists must be aligned to 8 bytes */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2, &chip->bdbars) < 0) { snd_intel8x0m_free(chip); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 0d81eac0a478..2b8204a13c69 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2275,7 +2275,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->idRegPtr, stateName[korg1212->cardState]); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) { snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%zd bytes)\n", sizeof(struct KorgSharedBuffer)); snd_korg1212_free(korg1212); @@ -2290,7 +2290,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->DataBufsSize = sizeof(struct KorgAudioBuffer) * kNumBuffers; - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, korg1212->DataBufsSize, &korg1212->dma_play) < 0) { snd_printk(KERN_ERR "korg1212: can not allocate play data buffer memory (%d bytes)\n", korg1212->DataBufsSize); snd_korg1212_free(korg1212); @@ -2302,7 +2302,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, K1212_DEBUG_PRINTK("K1212_DEBUG: Play Data Area = 0x%p (0x%08x), %d bytes\n", korg1212->playDataBufsPtr, korg1212->PlayDataPhy, korg1212->DataBufsSize); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, korg1212->DataBufsSize, &korg1212->dma_rec) < 0) { snd_printk(KERN_ERR "korg1212: can not allocate record data buffer memory (%d bytes)\n", korg1212->DataBufsSize); snd_korg1212_free(korg1212); @@ -2337,7 +2337,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, return err; } - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, dsp_code->size, &korg1212->dma_dsp) < 0) { snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size); snd_korg1212_free(korg1212); diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 5cda3488ceab..21ac9d003e8e 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -350,7 +350,7 @@ static int setup_corb_rirb(struct lola *chip) unsigned long end_time; err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, PAGE_SIZE, &chip->rb); if (err < 0) return err; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index f032d7ceb6e5..856bcca60128 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -591,7 +591,7 @@ int lola_create_pcm(struct lola *chip) for (i = 0; i < 2; i++) { err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, PAGE_SIZE, &chip->pcm[i].bdl); if (err < 0) return err; @@ -611,7 +611,7 @@ int lola_create_pcm(struct lola *chip) } /* buffer pre-allocation */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 1024 * 64, 32 * 1024 * 1024); return 0; } diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index fe10714380f2..d0f63fa54121 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -846,7 +846,7 @@ static int lx_pcm_create(struct lx6464es *chip) strcpy(pcm->name, card_name); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, size, size); chip->pcm = pcm; diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 19fa73df0846..cc8594d76c70 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -1861,7 +1861,8 @@ snd_m3_pcm(struct snd_m3 * chip, int device) chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 64*1024); + &chip->pci->dev, + 64*1024, 64*1024); return 0; } diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index e5279ce54ee1..674d37ec96b3 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -948,7 +948,8 @@ static void preallocate_buffers(struct snd_mixart *chip, struct snd_pcm *pcm) } #endif snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->mgr->pci), 32*1024, 32*1024); + &chip->mgr->pci->dev, + 32*1024, 32*1024); } /* @@ -1360,7 +1361,7 @@ static int snd_mixart_probe(struct pci_dev *pci, /* create array of streaminfo */ size = PAGE_ALIGN( (MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS * sizeof(struct mixart_flowinfo)) ); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, size, &mgr->flowinfo) < 0) { snd_mixart_free(mgr); return -ENOMEM; @@ -1371,7 +1372,7 @@ static int snd_mixart_probe(struct pci_dev *pci, /* create array of bufferinfo */ size = PAGE_ALIGN( (MIXART_MAX_STREAM_PER_CARD * MIXART_MAX_CARDS * sizeof(struct mixart_bufferinfo)) ); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, size, &mgr->bufferinfo) < 0) { snd_mixart_free(mgr); return -ENOMEM; diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index e6aa16646fd4..203c8fe48a01 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -713,13 +713,13 @@ int oxygen_pcm_init(struct oxygen *chip) if (outs) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, DEFAULT_BUFFER_BYTES_MULTICH, BUFFER_BYTES_MAX_MULTICH); if (ins) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, DEFAULT_BUFFER_BYTES, BUFFER_BYTES_MAX); } @@ -739,7 +739,7 @@ int oxygen_pcm_init(struct oxygen *chip) pcm->private_data = chip; strcpy(pcm->name, "Digital"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, DEFAULT_BUFFER_BYTES, BUFFER_BYTES_MAX); } @@ -769,7 +769,7 @@ int oxygen_pcm_init(struct oxygen *chip) pcm->private_data = chip; strcpy(pcm->name, outs ? "Front Panel" : "Analog 2"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, DEFAULT_BUFFER_BYTES, BUFFER_BYTES_MAX); } @@ -787,7 +787,7 @@ int oxygen_pcm_init(struct oxygen *chip) pcm->private_data = chip; strcpy(pcm->name, "Analog 3"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, DEFAULT_BUFFER_BYTES, BUFFER_BYTES_MAX); } diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index e493962d8455..4af34d6d92df 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1171,7 +1171,7 @@ int pcxhr_create_pcm(struct snd_pcxhr *chip) strcpy(pcm->name, name); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->mgr->pci), + &chip->mgr->pci->dev, 32*1024, 32*1024); chip->pcm = pcm; return 0; @@ -1644,7 +1644,7 @@ static int pcxhr_probe(struct pci_dev *pci, /* create hostport purgebuffer */ size = PAGE_ALIGN(sizeof(struct pcxhr_hostport)); - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, size, &mgr->hostport) < 0) { pcxhr_free(mgr); return -ENOMEM; diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index ac10d6f229d8..abcea86045ec 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1550,7 +1550,7 @@ snd_riptide_hw_params(struct snd_pcm_substream *substream, if (sgdlist->area) snd_dma_free_pages(sgdlist); if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, sizeof(struct sgd) * (DESC_MAX_MASK + 1), sgdlist)) < 0) { snd_printk(KERN_ERR "Riptide: failed to alloc %d dma bytes\n", @@ -1693,7 +1693,7 @@ static int snd_riptide_pcm(struct snd_riptide *chip, int device) strcpy(pcm->name, "RIPTIDE"); chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64 * 1024, 128 * 1024); return 0; } diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index b0b5e74e776c..ef7dd290ae05 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -905,7 +905,8 @@ static int sis_pcm_create(struct sis7019 *sis) * world if this fails. */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + &sis->pci->dev, + 64*1024, 128*1024); return 0; } diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 13103f5c309b..31cbc811ad37 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -884,7 +884,8 @@ static int snd_sonicvibes_pcm(struct sonicvibes *sonic, int device) sonic->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(sonic->pci), 64*1024, 128*1024); + &sonic->pci->dev, + 64*1024, 128*1024); return 0; } diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 8ac6f541b008..07022c0dad40 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -2184,14 +2184,16 @@ int snd_trident_pcm(struct snd_trident *trident, int device) struct snd_pcm_substream *substream; for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(trident->pci), + &trident->pci->dev, 64*1024, 128*1024); snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, - SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), + SNDRV_DMA_TYPE_DEV, + &trident->pci->dev, 64*1024, 128*1024); } else { snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(trident->pci), 64*1024, 128*1024); + &trident->pci->dev, + 64*1024, 128*1024); } return 0; @@ -2241,10 +2243,12 @@ int snd_trident_foldback_pcm(struct snd_trident *trident, int device) if (trident->tlb.entries) snd_pcm_lib_preallocate_pages_for_all(foldback, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(trident->pci), 0, 128*1024); + &trident->pci->dev, + 0, 128*1024); else snd_pcm_lib_preallocate_pages_for_all(foldback, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(trident->pci), 64*1024, 128*1024); + &trident->pci->dev, + 64*1024, 128*1024); return 0; } @@ -2278,7 +2282,9 @@ int snd_trident_spdif_pcm(struct snd_trident *trident, int device) strcpy(spdif->name, "Trident 4DWave IEC958"); trident->spdif = spdif; - snd_pcm_lib_preallocate_pages_for_all(spdif, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), 64*1024, 128*1024); + snd_pcm_lib_preallocate_pages_for_all(spdif, SNDRV_DMA_TYPE_DEV, + &trident->pci->dev, + 64*1024, 128*1024); return 0; } @@ -3336,7 +3342,7 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident) /* TLB array must be aligned to 16kB !!! so we allocate 32kB region and correct offset when necessary */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &trident->pci->dev, 2 * SNDRV_TRIDENT_MAX_PAGES * 4, &trident->tlb.buffer) < 0) { dev_err(trident->card->dev, "unable to allocate TLB buffer\n"); return -ENOMEM; @@ -3351,7 +3357,7 @@ static int snd_trident_tlb_alloc(struct snd_trident *trident) return -ENOMEM; /* allocate and setup silent page and initialise TLB entries */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(trident->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &trident->pci->dev, SNDRV_TRIDENT_PAGE_SIZE, &trident->tlb.silent_page) < 0) { dev_err(trident->card->dev, "unable to allocate silent page\n"); return -ENOMEM; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 902409dbc4c8..30c817b6b635 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -419,7 +419,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* the start of each lists must be aligned to 8 bytes, * but the kernel pages are much bigger, so we don't care */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, PAGE_ALIGN(VIA_TABLE_SIZE * 2 * 8), &dev->table) < 0) return -ENOMEM; @@ -1454,7 +1454,7 @@ static int snd_via8233_pcm_new(struct via82xx *chip) init_viadev(chip, chip->capture_devno, VIA_REG_CAPTURE_8233_STATUS, 6, 1); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, VIA_MAX_BUFSIZE); err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -1478,7 +1478,7 @@ static int snd_via8233_pcm_new(struct via82xx *chip) init_viadev(chip, chip->capture_devno + 1, VIA_REG_CAPTURE_8233_STATUS + 0x10, 7, 1); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, VIA_MAX_BUFSIZE); err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -1521,7 +1521,7 @@ static int snd_via8233a_pcm_new(struct via82xx *chip) init_viadev(chip, chip->capture_devno, VIA_REG_CAPTURE_8233_STATUS, 6, 1); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, VIA_MAX_BUFSIZE); err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -1547,7 +1547,7 @@ static int snd_via8233a_pcm_new(struct via82xx *chip) init_viadev(chip, chip->playback_devno, 0x30, 3, 0); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, VIA_MAX_BUFSIZE); return 0; } @@ -1577,7 +1577,7 @@ static int snd_via686_pcm_new(struct via82xx *chip) init_viadev(chip, 1, VIA_REG_CAPTURE_STATUS, 0, 1); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, VIA_MAX_BUFSIZE); return 0; } diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 2c8370cf03b3..0edb9ea6e8a6 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -272,7 +272,7 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre /* the start of each lists must be aligned to 8 bytes, * but the kernel pages are much bigger, so we don't care */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, PAGE_ALIGN(VIA_TABLE_SIZE * 2 * 8), &dev->table) < 0) return -ENOMEM; @@ -850,7 +850,7 @@ static int snd_via686_pcm_new(struct via82xx_modem *chip) init_viadev(chip, 1, VIA_REG_MI_STATUS, 1); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, 128*1024); return 0; } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 90400ebb64af..125c11ed5064 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -587,7 +587,7 @@ static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int static int snd_ymfpci_ac3_init(struct snd_ymfpci *chip) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, 4096, &chip->ac3_tmp_base) < 0) return -ENOMEM; @@ -1149,7 +1149,8 @@ int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device) chip->pcm = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, snd_pcm_std_chmaps, 2, 0, NULL); @@ -1184,7 +1185,8 @@ int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device) chip->pcm2 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return 0; } @@ -1217,7 +1219,8 @@ int snd_ymfpci_pcm_spdif(struct snd_ymfpci *chip, int device) chip->pcm_spdif = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return 0; } @@ -1258,7 +1261,8 @@ int snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device) chip->pcm_4ch = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + &chip->pci->dev, + 64*1024, 256*1024); return snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, surround_map, 2, 0, NULL); @@ -2108,7 +2112,7 @@ static int snd_ymfpci_memalloc(struct snd_ymfpci *chip) chip->work_size; /* work_ptr must be aligned to 256 bytes, but it's already covered with the kernel page allocation mechanism */ - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, size, &chip->work_ptr) < 0) return -ENOMEM; ptr = chip->work_ptr.area; From b65f131b8f47b399a0635f028af3888f8c741ae1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 16:18:56 +0100 Subject: [PATCH 087/136] ALSA: docs: Remove snd_dma_pci_data() usage in documentation The macro will be removed soon later, so update the documentation to reflect that as well. Link: https://lore.kernel.org/r/20191105151856.10785-25-tiwai@suse.de Signed-off-by: Takashi Iwai --- .../sound/kernel-api/writing-an-alsa-driver.rst | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index ba008ce28029..dcb7940435d9 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1383,7 +1383,7 @@ shows only the skeleton, how to build up the PCM interfaces. /* pre-allocation of buffers */ /* NOTE: this may fail */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, 64*1024); return 0; } @@ -1470,7 +1470,7 @@ buffer. For the pre-allocation, simply call the following: :: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), + &chip->pci->dev, 64*1024, 64*1024); It will allocate a buffer up to 64kB as default. Buffer management @@ -3514,7 +3514,7 @@ bus). :: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(pci), size, max); + &pci->dev, size, max); where ``size`` is the byte size to be pre-allocated and the ``max`` is the maximum size to be changed via the ``prealloc`` proc file. The @@ -3700,14 +3700,14 @@ For creating the SG-buffer handler, call :c:func:`snd_pcm_lib_preallocate_pages()` or :c:func:`snd_pcm_lib_preallocate_pages_for_all()` with ``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI -pre-allocator. You need to pass ``snd_dma_pci_data(pci)``, where pci is +pre-allocator. You need to pass ``&pci->dev``, where pci is the :c:type:`struct pci_dev ` pointer of the chip as well. :: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(pci), size, max); + &pci->dev, size, max); The ``struct snd_sg_buf`` instance is created as ``substream->dma_private`` in turn. You can cast the pointer like: From fdea53fe5de532969a332d6e5e727f2ad8bf084d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Nov 2019 16:42:57 +0100 Subject: [PATCH 088/136] ALSA: timer: Limit max amount of slave instances The fuzzer tries to open the timer instances as much as possible, and this may cause a system hiccup easily. We've already introduced the cap for the max number of available instances for the h/w timers, and we should put such a limit also to the slave timers, too. This patch introduces the limit to the multiple opened slave timers. The upper limit is hard-coded to 1000 for now, which should suffice for any practical usages up to now. Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/timer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/core/timer.c b/sound/core/timer.c index 5c9fbf3f4340..219249627cc3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -74,6 +74,9 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); +#define MAX_SLAVE_INSTANCES 1000 +static int num_slaves; + static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); @@ -250,6 +253,10 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EINVAL; goto unlock; } + if (num_slaves >= MAX_SLAVE_INSTANCES) { + err = -EBUSY; + goto unlock; + } timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { err = -ENOMEM; @@ -259,6 +266,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); + num_slaves++; err = snd_timer_check_slave(timeri); if (err < 0) { snd_timer_close_locked(timeri); @@ -350,6 +358,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri) } list_del(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; /* force to stop the timer */ snd_timer_stop(timeri); From ebfc6de29ae8e48f91e280fd37539188b5252b38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2019 20:20:06 +0100 Subject: [PATCH 089/136] ALSA: timer: Unify master/slave linking code The code in both snd_timer_check_master() and snd_timer_check_slave() are almost identical, both check whether the master/slave link and does linkage. Factor out the common code and call it from both functions for readability. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20191107192008.32331-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/timer.c | 66 +++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 33 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 013f0e69ff0f..21c2d25e4074 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -163,6 +163,28 @@ static void snd_timer_request(struct snd_timer_id *tid) #endif +/* move the slave if it belongs to the master; return 1 if match */ +static int check_matching_master_slave(struct snd_timer_instance *master, + struct snd_timer_instance *slave) +{ + if (slave->slave_class != master->slave_class || + slave->slave_id != master->slave_id) + return 0; + if (master->timer->num_instances >= master->timer->max_instances) + return -EBUSY; + list_move_tail(&slave->open_list, &master->slave_list_head); + master->timer->num_instances++; + spin_lock_irq(&slave_active_lock); + spin_lock(&master->timer->lock); + slave->master = master; + slave->timer = master->timer; + if (slave->flags & SNDRV_TIMER_IFLG_RUNNING) + list_add_tail(&slave->active_list, &master->slave_active_head); + spin_unlock(&master->timer->lock); + spin_unlock_irq(&slave_active_lock); + return 1; +} + /* * look for a master instance matching with the slave id of the given slave. * when found, relink the open_link of the slave. @@ -173,27 +195,18 @@ static int snd_timer_check_slave(struct snd_timer_instance *slave) { struct snd_timer *timer; struct snd_timer_instance *master; + int err = 0; /* FIXME: it's really dumb to look up all entries.. */ list_for_each_entry(timer, &snd_timer_list, device_list) { list_for_each_entry(master, &timer->open_list_head, open_list) { - if (slave->slave_class == master->slave_class && - slave->slave_id == master->slave_id) { - if (master->timer->num_instances >= - master->timer->max_instances) - return -EBUSY; - list_move_tail(&slave->open_list, - &master->slave_list_head); - master->timer->num_instances++; - spin_lock_irq(&slave_active_lock); - slave->master = master; - slave->timer = master->timer; - spin_unlock_irq(&slave_active_lock); - return 0; - } + err = check_matching_master_slave(master, slave); + if (err != 0) /* match found or error */ + goto out; } } - return 0; + out: + return err < 0 ? err : 0; } /* @@ -205,28 +218,15 @@ static int snd_timer_check_slave(struct snd_timer_instance *slave) static int snd_timer_check_master(struct snd_timer_instance *master) { struct snd_timer_instance *slave, *tmp; + int err = 0; /* check all pending slaves */ list_for_each_entry_safe(slave, tmp, &snd_timer_slave_list, open_list) { - if (slave->slave_class == master->slave_class && - slave->slave_id == master->slave_id) { - if (master->timer->num_instances >= - master->timer->max_instances) - return -EBUSY; - list_move_tail(&slave->open_list, &master->slave_list_head); - master->timer->num_instances++; - spin_lock_irq(&slave_active_lock); - spin_lock(&master->timer->lock); - slave->master = master; - slave->timer = master->timer; - if (slave->flags & SNDRV_TIMER_IFLG_RUNNING) - list_add_tail(&slave->active_list, - &master->slave_active_head); - spin_unlock(&master->timer->lock); - spin_unlock_irq(&slave_active_lock); - } + err = check_matching_master_slave(master, slave); + if (err < 0) + break; } - return 0; + return err < 0 ? err : 0; } static int snd_timer_close_locked(struct snd_timer_instance *timeri, From 33bbb8a0ecd142fb5e78f26f3a0b927ae77c30e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2019 20:20:07 +0100 Subject: [PATCH 090/136] ALSA: timer: Make snd_timer_close() returning void The function doesn't return any useful value, so let's make it void to be clearer. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20191107192008.32331-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/timer.h | 2 +- sound/core/timer.c | 18 +++++++----------- 2 files changed, 8 insertions(+), 12 deletions(-) diff --git a/include/sound/timer.h b/include/sound/timer.h index 199c36295a0d..8a13c09c85f7 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -119,7 +119,7 @@ int snd_timer_global_free(struct snd_timer *timer); int snd_timer_global_register(struct snd_timer *timer); int snd_timer_open(struct snd_timer_instance **ti, char *owner, struct snd_timer_id *tid, unsigned int slave_id); -int snd_timer_close(struct snd_timer_instance *timeri); +void snd_timer_close(struct snd_timer_instance *timeri); unsigned long snd_timer_resolution(struct snd_timer_instance *timeri); int snd_timer_start(struct snd_timer_instance *timeri, unsigned int ticks); int snd_timer_stop(struct snd_timer_instance *timeri); diff --git a/sound/core/timer.c b/sound/core/timer.c index 21c2d25e4074..c0a73913ec62 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -229,8 +229,8 @@ static int snd_timer_check_master(struct snd_timer_instance *master) return err < 0 ? err : 0; } -static int snd_timer_close_locked(struct snd_timer_instance *timeri, - struct device **card_devp_to_put); +static void snd_timer_close_locked(struct snd_timer_instance *timeri, + struct device **card_devp_to_put); /* * open a timer instance @@ -351,8 +351,8 @@ EXPORT_SYMBOL(snd_timer_open); * close a timer instance * call this with register_mutex down. */ -static int snd_timer_close_locked(struct snd_timer_instance *timeri, - struct device **card_devp_to_put) +static void snd_timer_close_locked(struct snd_timer_instance *timeri, + struct device **card_devp_to_put) { struct snd_timer *timer = timeri->timer; struct snd_timer_instance *slave, *tmp; @@ -414,28 +414,24 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri, *card_devp_to_put = &timer->card->card_dev; module_put(timer->module); } - - return 0; } /* * close a timer instance */ -int snd_timer_close(struct snd_timer_instance *timeri) +void snd_timer_close(struct snd_timer_instance *timeri) { struct device *card_dev_to_put = NULL; - int err; if (snd_BUG_ON(!timeri)) - return -ENXIO; + return; mutex_lock(®ister_mutex); - err = snd_timer_close_locked(timeri, &card_dev_to_put); + snd_timer_close_locked(timeri, &card_dev_to_put); mutex_unlock(®ister_mutex); /* put_device() is called after unlock for avoiding deadlock */ if (card_dev_to_put) put_device(card_dev_to_put); - return err; } EXPORT_SYMBOL(snd_timer_close); From 6a34367e52caea1413eb0a0dcbb524f0c9b67e82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2019 20:20:08 +0100 Subject: [PATCH 091/136] ALSA: timer: Fix possible race at assigning a timer instance When a new timer instance is created and assigned to the active link in snd_timer_open(), the caller still doesn't (can't) set its callback and callback data. In both the user-timer and the sequencer-timer code, they do manually set up the callbacks after calling snd_timer_open(). This has a potential risk of race when the timer instance is added to the already running timer target, as the callback might get triggered during setting up the callback itself. This patch tries to address it by changing the API usage slightly: - An empty timer instance is created at first via the new function snd_timer_instance_new(). This object isn't linked to the timer list yet. - The caller sets up the callbacks and others stuff for the new timer instance. - The caller invokes snd_timer_open() with this instance, so that it's linked to the target timer. For closing, do similarly: - Call snd_timer_close(). This unlinks the timer instance from the timer list. - Free the timer instance via snd_timer_instance_free() after that. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20191107192008.32331-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/timer.h | 4 +- sound/core/seq/seq_timer.c | 18 ++++--- sound/core/timer.c | 101 ++++++++++++++++++------------------- 3 files changed, 65 insertions(+), 58 deletions(-) diff --git a/include/sound/timer.h b/include/sound/timer.h index 8a13c09c85f7..a53e37bcd746 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -118,7 +118,9 @@ int snd_timer_global_new(char *id, int device, struct snd_timer **rtimer); int snd_timer_global_free(struct snd_timer *timer); int snd_timer_global_register(struct snd_timer *timer); -int snd_timer_open(struct snd_timer_instance **ti, char *owner, struct snd_timer_id *tid, unsigned int slave_id); +struct snd_timer_instance *snd_timer_instance_new(const char *owner); +void snd_timer_instance_free(struct snd_timer_instance *timeri); +int snd_timer_open(struct snd_timer_instance *timeri, struct snd_timer_id *tid, unsigned int slave_id); void snd_timer_close(struct snd_timer_instance *timeri); unsigned long snd_timer_resolution(struct snd_timer_instance *timeri); int snd_timer_start(struct snd_timer_instance *timeri, unsigned int ticks); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 161f3170bd7e..63dc7bdb622d 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -272,7 +272,13 @@ int snd_seq_timer_open(struct snd_seq_queue *q) return -EINVAL; if (tmr->alsa_id.dev_class != SNDRV_TIMER_CLASS_SLAVE) tmr->alsa_id.dev_sclass = SNDRV_TIMER_SCLASS_SEQUENCER; - err = snd_timer_open(&t, str, &tmr->alsa_id, q->queue); + t = snd_timer_instance_new(str); + if (!t) + return -ENOMEM; + t->callback = snd_seq_timer_interrupt; + t->callback_data = q; + t->flags |= SNDRV_TIMER_IFLG_AUTO; + err = snd_timer_open(t, &tmr->alsa_id, q->queue); if (err < 0 && tmr->alsa_id.dev_class != SNDRV_TIMER_CLASS_SLAVE) { if (tmr->alsa_id.dev_class != SNDRV_TIMER_CLASS_GLOBAL || tmr->alsa_id.device != SNDRV_TIMER_GLOBAL_SYSTEM) { @@ -282,16 +288,14 @@ int snd_seq_timer_open(struct snd_seq_queue *q) tid.dev_sclass = SNDRV_TIMER_SCLASS_SEQUENCER; tid.card = -1; tid.device = SNDRV_TIMER_GLOBAL_SYSTEM; - err = snd_timer_open(&t, str, &tid, q->queue); + err = snd_timer_open(t, &tid, q->queue); } } if (err < 0) { pr_err("ALSA: seq fatal error: cannot create timer (%i)\n", err); + snd_timer_instance_free(t); return err; } - t->callback = snd_seq_timer_interrupt; - t->callback_data = q; - t->flags |= SNDRV_TIMER_IFLG_AUTO; spin_lock_irq(&tmr->lock); tmr->timeri = t; spin_unlock_irq(&tmr->lock); @@ -310,8 +314,10 @@ int snd_seq_timer_close(struct snd_seq_queue *q) t = tmr->timeri; tmr->timeri = NULL; spin_unlock_irq(&tmr->lock); - if (t) + if (t) { snd_timer_close(t); + snd_timer_instance_free(t); + } return 0; } diff --git a/sound/core/timer.c b/sound/core/timer.c index c0a73913ec62..9091030f4a89 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -88,12 +88,11 @@ static void snd_timer_reschedule(struct snd_timer * timer, unsigned long ticks_l /* * create a timer instance with the given owner string. - * when timer is not NULL, increments the module counter */ -static struct snd_timer_instance *snd_timer_instance_new(char *owner, - struct snd_timer *timer) +struct snd_timer_instance *snd_timer_instance_new(const char *owner) { struct snd_timer_instance *timeri; + timeri = kzalloc(sizeof(*timeri), GFP_KERNEL); if (timeri == NULL) return NULL; @@ -108,15 +107,20 @@ static struct snd_timer_instance *snd_timer_instance_new(char *owner, INIT_LIST_HEAD(&timeri->slave_list_head); INIT_LIST_HEAD(&timeri->slave_active_head); - timeri->timer = timer; - if (timer && !try_module_get(timer->module)) { - kfree(timeri->owner); - kfree(timeri); - return NULL; - } - return timeri; } +EXPORT_SYMBOL(snd_timer_instance_new); + +void snd_timer_instance_free(struct snd_timer_instance *timeri) +{ + if (timeri) { + if (timeri->private_free) + timeri->private_free(timeri); + kfree(timeri->owner); + kfree(timeri); + } +} +EXPORT_SYMBOL(snd_timer_instance_free); /* * find a timer instance from the given timer id @@ -236,12 +240,11 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, * open a timer instance * when opening a master, the slave id must be here given. */ -int snd_timer_open(struct snd_timer_instance **ti, - char *owner, struct snd_timer_id *tid, +int snd_timer_open(struct snd_timer_instance *timeri, + struct snd_timer_id *tid, unsigned int slave_id) { struct snd_timer *timer; - struct snd_timer_instance *timeri = NULL; struct device *card_dev_to_put = NULL; int err; @@ -259,22 +262,14 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EBUSY; goto unlock; } - timeri = snd_timer_instance_new(owner, NULL); - if (!timeri) { - err = -ENOMEM; - goto unlock; - } timeri->slave_class = tid->dev_sclass; timeri->slave_id = tid->device; timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); num_slaves++; err = snd_timer_check_slave(timeri); - if (err < 0) { - snd_timer_close_locked(timeri, &card_dev_to_put); - timeri = NULL; - } - goto unlock; + if (err < 0) + goto close; } /* open a master instance */ @@ -304,45 +299,40 @@ int snd_timer_open(struct snd_timer_instance **ti, err = -EBUSY; goto unlock; } - timeri = snd_timer_instance_new(owner, timer); - if (!timeri) { - err = -ENOMEM; + if (!try_module_get(timer->module)) { + err = -EBUSY; goto unlock; } /* take a card refcount for safe disconnection */ - if (timer->card) + if (timer->card) { get_device(&timer->card->card_dev); - timeri->slave_class = tid->dev_sclass; - timeri->slave_id = slave_id; + card_dev_to_put = &timer->card->card_dev; + } if (list_empty(&timer->open_list_head) && timer->hw.open) { err = timer->hw.open(timer); if (err) { - kfree(timeri->owner); - kfree(timeri); - timeri = NULL; - - if (timer->card) - card_dev_to_put = &timer->card->card_dev; module_put(timer->module); goto unlock; } } + timeri->timer = timer; + timeri->slave_class = tid->dev_sclass; + timeri->slave_id = slave_id; + list_add_tail(&timeri->open_list, &timer->open_list_head); timer->num_instances++; err = snd_timer_check_master(timeri); - if (err < 0) { + close: + if (err < 0) snd_timer_close_locked(timeri, &card_dev_to_put); - timeri = NULL; - } unlock: mutex_unlock(®ister_mutex); /* put_device() is called after unlock for avoiding deadlock */ - if (card_dev_to_put) + if (err < 0 && card_dev_to_put) put_device(card_dev_to_put); - *ti = timeri; return err; } EXPORT_SYMBOL(snd_timer_open); @@ -363,9 +353,11 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, spin_unlock_irq(&timer->lock); } - list_del(&timeri->open_list); - if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) - num_slaves--; + if (!list_empty(&timeri->open_list)) { + list_del_init(&timeri->open_list); + if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE) + num_slaves--; + } /* force to stop the timer */ snd_timer_stop(timeri); @@ -384,6 +376,7 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, /* remove slave links */ spin_lock_irq(&slave_active_lock); spin_lock(&timer->lock); + timeri->timer = NULL; list_for_each_entry_safe(slave, tmp, &timeri->slave_list_head, open_list) { list_move_tail(&slave->open_list, &snd_timer_slave_list); @@ -401,11 +394,6 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, timer = NULL; } - if (timeri->private_free) - timeri->private_free(timeri); - kfree(timeri->owner); - kfree(timeri); - if (timer) { if (list_empty(&timer->open_list_head) && timer->hw.close) timer->hw.close(timer); @@ -1480,8 +1468,10 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) tu = file->private_data; file->private_data = NULL; mutex_lock(&tu->ioctl_lock); - if (tu->timeri) + if (tu->timeri) { snd_timer_close(tu->timeri); + snd_timer_instance_free(tu->timeri); + } mutex_unlock(&tu->ioctl_lock); kfree(tu->queue); kfree(tu->tqueue); @@ -1722,6 +1712,7 @@ static int snd_timer_user_tselect(struct file *file, tu = file->private_data; if (tu->timeri) { snd_timer_close(tu->timeri); + snd_timer_instance_free(tu->timeri); tu->timeri = NULL; } if (copy_from_user(&tselect, _tselect, sizeof(tselect))) { @@ -1731,9 +1722,11 @@ static int snd_timer_user_tselect(struct file *file, sprintf(str, "application %i", current->pid); if (tselect.id.dev_class != SNDRV_TIMER_CLASS_SLAVE) tselect.id.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION; - err = snd_timer_open(&tu->timeri, str, &tselect.id, current->pid); - if (err < 0) + tu->timeri = snd_timer_instance_new(str); + if (!tu->timeri) { + err = -ENOMEM; goto __err; + } tu->timeri->flags |= SNDRV_TIMER_IFLG_FAST; tu->timeri->callback = tu->tread @@ -1742,6 +1735,12 @@ static int snd_timer_user_tselect(struct file *file, tu->timeri->callback_data = (void *)tu; tu->timeri->disconnect = snd_timer_user_disconnect; + err = snd_timer_open(tu->timeri, &tselect.id, current->pid); + if (err < 0) { + snd_timer_instance_free(tu->timeri); + tu->timeri = NULL; + } + __err: return err; } From 2406ff9b86aa1b77fe1a6d15f37195ac1fdb2a14 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 17:56:26 +0100 Subject: [PATCH 092/136] ALSA: pcm: Yet another missing check of non-cached buffer type For non-x86 architectures, SNDRV_DMA_TYPE_DEV_UC should be treated equivalent with SNDRV_DMA_TYPE_DEV, where the default mmap handler still checks only about SNDRV_DMA_TYPE_DEV. Make the check more proper. Note that all existing users of *_UC buffer types are x86-only, so this doesn't fix any bug, but just for consistency. Fixes: 42e748a0b325 ("ALSA: memalloc: Add non-cached buffer type") Link: https://lore.kernel.org/r/20191108165626.5947-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 998c63192ae4..0c27009dc3df 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3416,7 +3416,8 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, #endif /* CONFIG_GENERIC_ALLOCATOR */ #ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page && - substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV || + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_UC)) return dma_mmap_coherent(substream->dma_buffer.dev.dev, area, substream->runtime->dma_area, From 1a4860325cc740b4ec4b849f31fa694cf8870569 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:34 +0100 Subject: [PATCH 093/136] ASoC: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/au1x/dma.c | 2 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 49556eb409cd..054dfda89d3e 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -293,7 +293,7 @@ static int alchemy_pcm_new(struct snd_soc_component *component, struct snd_pcm *pcm = rtd->pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1); + NULL, 65536, (4096 * 1024) - 1); return 0; } diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c index de6fcc808832..4771eb5fbe2a 100644 --- a/sound/soc/dwc/dwc-pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -258,7 +258,7 @@ static int dw_pcm_new(struct snd_soc_component *component, snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), size, size); + NULL, size, size); return 0; } diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 93bb80d089be..2b0eca02a8b9 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -309,7 +309,7 @@ static int camelot_pcm_new(struct snd_soc_component *component, */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX); return 0; From 26105a6f05709ad7059c91e28005a5c75be24819 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:35 +0100 Subject: [PATCH 094/136] ASoC: rt5514-spi: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Since it requires the specific buffer type (SNDRV_DMA_TYPE_VMALLOC), it's set in the pcm_new ops now. Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/codecs/rt5514-spi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 57ff5aee452d..f1b7b947ecbd 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -219,8 +219,7 @@ static int rt5514_spi_hw_params(struct snd_soc_component *component, u8 buf[8]; mutex_lock(&rt5514_dsp->dma_lock); - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); rt5514_dsp->substream = substream; rt5514_dsp->dma_offset = 0; @@ -246,7 +245,7 @@ static int rt5514_spi_hw_free(struct snd_soc_component *component, cancel_delayed_work_sync(&rt5514_dsp->copy_work); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static snd_pcm_uframes_t rt5514_spi_pcm_pointer( @@ -260,12 +259,6 @@ static snd_pcm_uframes_t rt5514_spi_pcm_pointer( return bytes_to_frames(runtime, rt5514_dsp->dma_offset); } -static struct page *rt5514_spi_pcm_page(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned long offset) -{ - return snd_pcm_lib_get_vmalloc_page(substream, offset); -} static int rt5514_spi_pcm_probe(struct snd_soc_component *component) { @@ -298,6 +291,14 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) return 0; } +static int rt5514_spi_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); + return 0; +} + static const struct snd_soc_component_driver rt5514_spi_component = { .name = DRV_NAME, .probe = rt5514_spi_pcm_probe, @@ -305,7 +306,7 @@ static const struct snd_soc_component_driver rt5514_spi_component = { .hw_params = rt5514_spi_hw_params, .hw_free = rt5514_spi_hw_free, .pointer = rt5514_spi_pcm_pointer, - .page = rt5514_spi_pcm_page, + .pcm_construct = rt5514_spi_pcm_new, }; /** From 1508f32f1adb220e61d1b7839e35d93ebee89f58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:36 +0100 Subject: [PATCH 095/136] ASoC: intel: Drop superfluous snd_pcm_sgbuf_ops_page snd_pcm_sgbuf_ops_page is no longer needed to be set explicitly to PCM page ops since the recent change in the PCM core (*). Leaving it NULL should work as long as the preallocation has been done properly. This patch drops the redundant lines. (*) 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/intel/haswell/sst-haswell-pcm.c | 12 ------------ sound/soc/intel/skylake/skl-pcm.c | 12 ------------ 2 files changed, 24 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 6e498a581d20..a3a5bba2fbd9 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -796,17 +796,6 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_soc_component *component, return offset; } -#ifdef CONFIG_SND_DMA_SGBUF -static struct page *hsw_pcm_page(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned long offset) -{ - return snd_pcm_sgbuf_ops_page(substream, offset); -} -#else -#define hsw_pcm_page NULL -#endif /* CONFIG_SND_DMA_SGBUF */ - static int hsw_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -1128,7 +1117,6 @@ static const struct snd_soc_component_driver hsw_dai_component = { .hw_free = hsw_pcm_hw_free, .trigger = hsw_pcm_trigger, .pointer = hsw_pcm_pointer, - .page = hsw_pcm_page, .ioctl = snd_soc_pcm_lib_ioctl, .pcm_construct = hsw_pcm_new, .controls = hsw_volume_controls, diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 0850141c7d47..6bd14f65fe71 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1235,17 +1235,6 @@ static int skl_platform_soc_mmap(struct snd_soc_component *component, return snd_pcm_lib_default_mmap(substream, area); } -#ifdef CONFIG_SND_DMA_SGBUF -static struct page *skl_platform_soc_page(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned long offset) -{ - return snd_pcm_sgbuf_ops_page(substream, offset); -} -#else -#define skl_platform_soc_page NULL -#endif /* CONFIG_SND_DMA_SGBUF */ - static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { @@ -1493,7 +1482,6 @@ static const struct snd_soc_component_driver skl_component = { .pointer = skl_platform_soc_pointer, .get_time_info = skl_platform_soc_get_time_info, .mmap = skl_platform_soc_mmap, - .page = skl_platform_soc_page, .pcm_construct = skl_platform_soc_new, .pcm_destruct = skl_platform_soc_free, .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ From 6420c24a30006e2fdbc292f88d37db56ca28f6e8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:37 +0100 Subject: [PATCH 096/136] ASoC: intel: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/intel/skylake/skl-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 6bd14f65fe71..8b9abb79a69e 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1314,7 +1314,7 @@ static int skl_platform_soc_new(struct snd_soc_component *component, size = MAX_PREALLOC_SIZE; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - snd_dma_pci_data(skl->pci), + &skl->pci->dev, size, MAX_PREALLOC_SIZE); } From b01aa131fae9da3c1898baff17176307c0e23e12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:38 +0100 Subject: [PATCH 097/136] ASoC: SOF: Drop superfluous snd_pcm_sgbuf_ops_page snd_pcm_sgbuf_ops_page is no longer needed to be set explicitly to PCM page ops since the recent change in the PCM core (*). Leaving it NULL should work as long as the preallocation has been done properly. This patch drops the redundant lines. (*) 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/sof/pcm.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 3d5cd1b445ba..549238a98b2a 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -454,17 +454,6 @@ static snd_pcm_uframes_t sof_pcm_pointer(struct snd_soc_component *component, return host; } -#ifdef CONFIG_SND_DMA_SGBUF -static struct page *sof_pcm_page(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned long offset) -{ - return snd_pcm_sgbuf_ops_page(substream, offset); -} -#else -#define sof_pcm_page NULL -#endif /* CONFIG_SND_DMA_SGBUF */ - static int sof_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -788,7 +777,6 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) pd->hw_free = sof_pcm_hw_free; pd->trigger = sof_pcm_trigger; pd->pointer = sof_pcm_pointer; - pd->page = sof_pcm_page; #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) pd->compr_ops = &sof_compressed_ops; From f541220cd590e4a7436913d6f7d41ff4aba67982 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:39 +0100 Subject: [PATCH 098/136] ASoC: rt5677-spi: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Acked-by: Mark Brown Acked-by: Curtis Malainey Link: https://lore.kernel.org/r/20191108094641.20086-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/codecs/rt5677-spi.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 3aa3ea7c2768..0c90f99b693f 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -128,8 +128,7 @@ static int rt5677_spi_hw_params( int ret; mutex_lock(&rt5677_dsp->dma_lock); - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); rt5677_dsp->substream = substream; mutex_unlock(&rt5677_dsp->dma_lock); @@ -147,7 +146,7 @@ static int rt5677_spi_hw_free( rt5677_dsp->substream = NULL; mutex_unlock(&rt5677_dsp->dma_lock); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static int rt5677_spi_prepare( @@ -361,12 +360,12 @@ static void rt5677_spi_copy_work(struct work_struct *work) mutex_unlock(&rt5677_dsp->dma_lock); } -static struct page *rt5677_spi_pcm_page( - struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned long offset) +static int rt5677_spi_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) { - return snd_pcm_lib_get_vmalloc_page(substream, offset); + snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); + return 0; } static int rt5677_spi_pcm_probe(struct snd_soc_component *component) @@ -394,7 +393,7 @@ static const struct snd_soc_component_driver rt5677_spi_dai_component = { .hw_free = rt5677_spi_hw_free, .prepare = rt5677_spi_prepare, .pointer = rt5677_spi_pcm_pointer, - .page = rt5677_spi_pcm_page, + .pcm_construct = rt5677_spi_pcm_new, }; /* Select a suitable transfer command for the next transfer to ensure From e610748ad4388f02a2cd8b3ad051a662c03dbfbc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:40 +0100 Subject: [PATCH 099/136] ASoC: cros_ec_codec: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/codecs/cros_ec_codec.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index dd14caf9091a..7b17f39a6a10 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -918,8 +918,7 @@ static int wov_pcm_hw_params(struct snd_soc_component *component, priv->wov_burst_read = true; mutex_unlock(&priv->wov_dma_lock); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } static int wov_pcm_hw_free(struct snd_soc_component *component, @@ -935,7 +934,7 @@ static int wov_pcm_hw_free(struct snd_soc_component *component, cancel_delayed_work_sync(&priv->wov_copy_work); - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } static snd_pcm_uframes_t wov_pcm_pointer(struct snd_soc_component *component, @@ -948,11 +947,12 @@ static snd_pcm_uframes_t wov_pcm_pointer(struct snd_soc_component *component, return bytes_to_frames(runtime, priv->wov_dma_offset); } -static struct page *wov_pcm_page(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - unsigned long offset) +static int wov_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) { - return snd_pcm_lib_get_vmalloc_page(substream, offset); + snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); + return 0; } static const struct snd_soc_component_driver wov_component_driver = { @@ -964,7 +964,7 @@ static const struct snd_soc_component_driver wov_component_driver = { .hw_params = wov_pcm_hw_params, .hw_free = wov_pcm_hw_free, .pointer = wov_pcm_pointer, - .page = wov_pcm_page, + .pcm_construct = wov_pcm_new, }; static int cros_ec_codec_platform_probe(struct platform_device *pdev) From fc7af6bc2707e32d73c639d63f1543d890a00db4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 8 Nov 2019 10:46:41 +0100 Subject: [PATCH 100/136] ALSA: pcm: Unexport snd_pcm_sgbuf_ops_page The helper is no longer referred after the recent code refactoring. Drop the export for saving some bits and future misuse. Acked-by: Mark Brown Link: https://lore.kernel.org/r/20191108094641.20086-9-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 8 -------- sound/core/pcm_local.h | 5 +++++ sound/core/pcm_memory.c | 3 +-- 3 files changed, 6 insertions(+), 10 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index bbe6eb1ff5d2..2c0aa884f5f1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1236,14 +1236,6 @@ static inline int snd_pcm_lib_alloc_vmalloc_32_buffer */ #define snd_pcm_substream_sgbuf(substream) \ snd_pcm_get_dma_buf(substream)->private_data - -struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, - unsigned long offset); -#else /* !SND_DMA_SGBUF */ -/* - * fake using a continuous buffer - */ -#define snd_pcm_sgbuf_ops_page NULL #endif /* SND_DMA_SGBUF */ /** diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 1161ab2d6a5b..5565e1c4196a 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -67,4 +67,9 @@ static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} void __snd_pcm_xrun(struct snd_pcm_substream *substream); void snd_pcm_group_init(struct snd_pcm_group *group); +#ifdef CONFIG_SND_DMA_SGBUF +struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, + unsigned long offset); +#endif + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 17ee361ce791..286f333f8e4c 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -255,7 +255,7 @@ void snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); #ifdef CONFIG_SND_DMA_SGBUF -/** +/* * snd_pcm_sgbuf_ops_page - get the page struct at the given offset * @substream: the pcm substream instance * @offset: the buffer offset @@ -273,7 +273,6 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigne return NULL; return sgbuf->page_table[idx]; } -EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page); #endif /* CONFIG_SND_DMA_SGBUF */ /** From 67b2945d632328fa0245279e8281e34bf1c78ae2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2019 14:36:17 +0100 Subject: [PATCH 101/136] usb: gadget: u_audio: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINUOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Acked-by: Greg Kroah-Hartman Link: https://lore.kernel.org/r/20191108164214.611-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- drivers/usb/gadget/function/u_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/usb/gadget/function/u_audio.c b/drivers/usb/gadget/function/u_audio.c index 56906d15fb55..7ec6a996af26 100644 --- a/drivers/usb/gadget/function/u_audio.c +++ b/drivers/usb/gadget/function/u_audio.c @@ -585,7 +585,7 @@ int g_audio_setup(struct g_audio *g_audio, const char *pcm_name, sprintf(card->longname, "%s %i", card_name, card->dev->id); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), 0, BUFF_SIZE_MAX); + NULL, 0, BUFF_SIZE_MAX); err = snd_card_register(card); From 9cb7831384cefdb91378738b66c7f3e1bfd7e26c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2019 16:16:57 +0100 Subject: [PATCH 102/136] staging: most: Convert to the common vmalloc memalloc The recent change (*) in the ALSA memalloc core allows us to drop the special vmalloc-specific allocation and page handling. This patch coverts to the common code. (*) 1fe7f397cfe2: ALSA: memalloc: Add vmalloc buffer allocation support 7e8edae39fd1: ALSA: pcm: Handle special page mapping in the default mmap handler Acked-by: Greg Kroah-Hartman Link: https://lore.kernel.org/r/20191108164528.998-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- drivers/staging/most/sound/sound.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/staging/most/sound/sound.c b/drivers/staging/most/sound/sound.c index 79817061fcfa..add16997f129 100644 --- a/drivers/staging/most/sound/sound.c +++ b/drivers/staging/most/sound/sound.c @@ -344,8 +344,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, pr_err("Requested number of channels not supported.\n"); return -EINVAL; } - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } /** @@ -359,7 +358,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, */ static int pcm_hw_free(struct snd_pcm_substream *substream) { - return snd_pcm_lib_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_pages(substream); } /** @@ -469,7 +468,6 @@ static const struct snd_pcm_ops pcm_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, }; static int split_arg_list(char *buf, u16 *ch_num, char **sample_res) @@ -663,6 +661,8 @@ static int audio_probe_channel(struct most_interface *iface, int channel_id, pcm->private_data = channel; strscpy(pcm->name, device_name, sizeof(pcm->name)); snd_pcm_set_ops(pcm, direction, &pcm_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); return 0; From ef99d99b2fd921148dc15e706f2fb071ba3fee6c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Nov 2019 14:33:58 +0100 Subject: [PATCH 103/136] media: solo6x10: Remove superfluous snd_dma_continuous_data() The recent change (commit 08422d2c559d: "ALSA: memalloc: Allow NULL device for SNDRV_DMA_TYPE_CONTINOUS type") made the PCM preallocation helper accepting NULL as the device pointer for the default usage. Drop the snd_dma_continuous_data() usage that became superfluous from the callers. Acked-by: Hans Verkuil Signed-off-by: Takashi Iwai --- drivers/media/pci/solo6x10/solo6x10-g723.c | 2 +- drivers/media/usb/usbtv/usbtv-audio.c | 3 +-- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/drivers/media/pci/solo6x10/solo6x10-g723.c b/drivers/media/pci/solo6x10/solo6x10-g723.c index 30c8f2ec9c3c..eaa57d835ea8 100644 --- a/drivers/media/pci/solo6x10/solo6x10-g723.c +++ b/drivers/media/pci/solo6x10/solo6x10-g723.c @@ -353,7 +353,7 @@ static int solo_snd_pcm_init(struct solo_dev *solo_dev) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + NULL, G723_PERIOD_BYTES * PERIODS, G723_PERIOD_BYTES * PERIODS); diff --git a/drivers/media/usb/usbtv/usbtv-audio.c b/drivers/media/usb/usbtv/usbtv-audio.c index 6f108996142d..e746c8ddfc49 100644 --- a/drivers/media/usb/usbtv/usbtv-audio.c +++ b/drivers/media/usb/usbtv/usbtv-audio.c @@ -378,8 +378,7 @@ int usbtv_audio_init(struct usbtv *usbtv) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usbtv_pcm_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), USBTV_AUDIO_BUFFER, - USBTV_AUDIO_BUFFER); + NULL, USBTV_AUDIO_BUFFER, USBTV_AUDIO_BUFFER); rv = snd_card_register(card); if (rv) From 3b316a8f6077a0003331aae43f02279ea62d54f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 08:01:47 +0100 Subject: [PATCH 104/136] media: tw686x: audio: Avoid non-standard macro usage Pass the device pointer from the PCI pointer directly, instead of a non-standard macro. The macro didn't give any better readability. Acked-by: Hans Verkuil Signed-off-by: Takashi Iwai --- drivers/media/pci/tw686x/tw686x-audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/media/pci/tw686x/tw686x-audio.c b/drivers/media/pci/tw686x/tw686x-audio.c index 40373bd23381..7786e51d19ae 100644 --- a/drivers/media/pci/tw686x/tw686x-audio.c +++ b/drivers/media/pci/tw686x/tw686x-audio.c @@ -300,7 +300,7 @@ static int tw686x_snd_pcm_init(struct tw686x_dev *dev) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(dev->pci_dev), + &dev->pci_dev->dev, TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX, TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX); return 0; From 2bc1e1fdd0fc54ef6aee14c7710af21ce816b398 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Nov 2019 08:14:59 +0100 Subject: [PATCH 105/136] ALSA: memalloc: Drop snd_dma_pci_data() macro Now that all users have been converted, we can drop snd_dma_pci_data() macro definition. Link: https://lore.kernel.org/r/20191108164637.1110-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 6ada3b8ede4e..3b47832b1c1f 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -21,7 +21,6 @@ struct snd_dma_device { struct device *dev; /* generic device */ }; -#define snd_dma_pci_data(pci) (&(pci)->dev) #define snd_dma_continuous_data(x) ((struct device *)(__force unsigned long)(x)) From 0c4f09ceec3d7afbef6ad4077d96e36e0fad4028 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Nov 2019 18:36:42 +0100 Subject: [PATCH 106/136] ALSA: timer: Fix the breakage of slave link open A silly mistake was made while applying the fix for potential races in commit 6a34367e52ca ("ALSA: timer: Fix possible race at assigning a timer instance"): when a slave PCM is opened and succeeds, it doesn't return but proceeds to the master timer open code instead. Plug the hole and beautify a bit. Fixes: 6a34367e52ca ("ALSA: timer: Fix possible race at assigning a timer instance") Reported-by: syzbot+4476917c053f60112c99@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20191111173642.6093-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/timer.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 9091030f4a89..24fed5c78273 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -268,8 +268,7 @@ int snd_timer_open(struct snd_timer_instance *timeri, list_add_tail(&timeri->open_list, &snd_timer_slave_list); num_slaves++; err = snd_timer_check_slave(timeri); - if (err < 0) - goto close; + goto list_added; } /* open a master instance */ @@ -324,7 +323,7 @@ int snd_timer_open(struct snd_timer_instance *timeri, list_add_tail(&timeri->open_list, &timer->open_list_head); timer->num_instances++; err = snd_timer_check_master(timeri); - close: +list_added: if (err < 0) snd_timer_close_locked(timeri, &card_dev_to_put); From 509526706e68e4fb47519f02f78b48c818869d4e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Nov 2019 15:32:43 +0100 Subject: [PATCH 107/136] ALSA: au88x0: Fix incorrect device pointer for preallocation The code change in commit 6974f8ad4494 ("ALSA: pci: Avoid non-standard macro usage") contained an incorrect conversion, which left the invalid pointer passed to the allocator for au88x0 driver. Fix it. Fixes: 6974f8ad4494 ("ALSA: pci: Avoid non-standard macro usage") Link: https://lore.kernel.org/r/20191112143243.22216-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 1012e8da29e7..a2dcf43beedf 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -637,7 +637,7 @@ static int snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) /* pre-allocation of Scatter-Gather buffers */ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - &chip->pci_dev, + &chip->pci_dev->dev, 0x10000, 0x10000); switch (VORTEX_PCM_TYPE(pcm)) { From cb45722b289b54476b68883985c2824c69a7fba9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Nov 2019 20:09:37 +0100 Subject: [PATCH 108/136] ALSA: hda/hdmi - Clean up Intel platform-specific fixup checks Introduce a new flag in hdmi_spec to indicate the Intel platform- specific fixups so that we can get rid of the lengthy codec ID checks. The flag is set in intel_hsw_common_init() commonly. Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20191111190937.19186-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 27 +++++---------------------- 1 file changed, 5 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d24bd95c6e95..69d1a6e41f0d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -37,25 +37,6 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); -#define is_haswell(codec) ((codec)->core.vendor_id == 0x80862807) -#define is_broadwell(codec) ((codec)->core.vendor_id == 0x80862808) -#define is_skylake(codec) ((codec)->core.vendor_id == 0x80862809) -#define is_broxton(codec) ((codec)->core.vendor_id == 0x8086280a) -#define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) -#define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ - ((codec)->core.vendor_id == 0x80862800)) -#define is_cannonlake(codec) ((codec)->core.vendor_id == 0x8086280c) -#define is_icelake(codec) ((codec)->core.vendor_id == 0x8086280f) -#define is_tigerlake(codec) ((codec)->core.vendor_id == 0x80862812) -#define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ - || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec) || is_geminilake(codec) \ - || is_cannonlake(codec) || is_icelake(codec) \ - || is_tigerlake(codec)) -#define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) -#define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) -#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) - struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -162,6 +143,7 @@ struct hdmi_spec { bool dyn_pin_out; bool dyn_pcm_assign; + bool intel_hsw_fixup; /* apply Intel platform-specific fixups */ /* * Non-generic VIA/NVIDIA specific */ @@ -925,7 +907,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return err; } - if (is_haswell_plus(codec)) { + if (spec->intel_hsw_fixup) { /* * on recent platforms IEC Coding Type is required for HBR @@ -1709,7 +1691,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) * To simplify the implementation, malloc all * the virtual pins in the initialization statically */ - if (is_haswell_plus(codec)) { + if (spec->intel_hsw_fixup) { /* * On Intel platforms, device entries number is * changed dynamically. If there is a DP MST @@ -1758,7 +1740,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin->dev_id = i; per_pin->non_pcm = false; snd_hda_set_dev_select(codec, pin_nid, i); - if (is_haswell_plus(codec)) + if (spec->intel_hsw_fixup) intel_haswell_fixup_connect_list(codec, pin_nid); err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) @@ -2825,6 +2807,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, spec->vendor_nid = vendor_nid; spec->port_map = port_map; spec->port_num = port_num; + spec->intel_hsw_fixup = true; intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); From f35ef592477c5347b8f780a5f0d4970671e22c61 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Nov 2019 17:56:13 +0100 Subject: [PATCH 109/136] ALSA: usb-audio: Add skip_validation option The unit descriptor validation may lead to a probe error when the device provides a buggy descriptor or the validator detected incorrectly. For identifying such an error and band-aiding, give a new module option, skip_validation. With this option, the driver ignores the validation errors with the hexdump of the unit descriptor, so we can check it in a bit more details. Link: https://lore.kernel.org/r/20191114165613.7422-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.c | 3 +++ sound/usb/usbaudio.h | 1 + sound/usb/validate.c | 23 ++++++++++++++++++++--- 3 files changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index db91dc76cc91..9f743ebae615 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -74,6 +74,7 @@ static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; bool snd_usb_use_vmalloc = true; +bool snd_usb_skip_validation; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); @@ -96,6 +97,8 @@ module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); +module_param_named(skip_validation, snd_usb_skip_validation, bool, 0444); +MODULE_PARM_DESC(skip_validation, "Skip unit descriptor validation (default: no)."); /* * we keep the snd_usb_audio_t instances by ourselves for merging diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index feb30f9c1716..ff3cbf653de8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -120,5 +120,6 @@ int snd_usb_lock_shutdown(struct snd_usb_audio *chip); void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); extern bool snd_usb_use_vmalloc; +extern bool snd_usb_skip_validation; #endif /* __USBAUDIO_H */ diff --git a/sound/usb/validate.c b/sound/usb/validate.c index 389e8657434a..36ae78c3da3d 100644 --- a/sound/usb/validate.c +++ b/sound/usb/validate.c @@ -322,11 +322,28 @@ static bool validate_desc(unsigned char *hdr, int protocol, bool snd_usb_validate_audio_desc(void *p, int protocol) { - return validate_desc(p, protocol, audio_validators); + unsigned char *c = p; + bool valid; + + valid = validate_desc(p, protocol, audio_validators); + if (!valid && snd_usb_skip_validation) { + print_hex_dump(KERN_ERR, "USB-audio: buggy audio desc: ", + DUMP_PREFIX_NONE, 16, 1, c, c[0], true); + valid = true; + } + return valid; } bool snd_usb_validate_midi_desc(void *p) { - return validate_desc(p, UAC_VERSION_1, midi_validators); -} + unsigned char *c = p; + bool valid; + valid = validate_desc(p, UAC_VERSION_1, midi_validators); + if (!valid && snd_usb_skip_validation) { + print_hex_dump(KERN_ERR, "USB-audio: buggy midi desc: ", + DUMP_PREFIX_NONE, 16, 1, c, c[0], true); + valid = true; + } + return valid; +} From d577cf76fa3192f9e656bb101aa54d25a6a71347 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 15 Nov 2019 14:44:47 +0200 Subject: [PATCH 110/136] ALSA: hda: hdmi - fix port numbering for ICL and TGL platforms Semantics of port#0 differ between ICL and TGL: ICL port#0 -> never used for HDAudio ICL port#1 -> should be mapped to first pin (0x04) TGL port#0 -> typically not used, but HW has the support, so should be mapped to first pin (0x04) TGL port#1 -> should be mapped to 2nd pin (0x06) Refactor the port mapping logic to allow to take the above differences into account. Fixes issues with HDAudio on some TGL platforms. Co-developed-by: Pan Xiuli Signed-off-by: Pan Xiuli Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20191115124449.20512-2-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 69d1a6e41f0d..dec90f931bae 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2664,7 +2664,7 @@ static int intel_pin2port(void *audio_ptr, int pin_nid) */ for (i = 0; i < spec->port_num; i++) { if (pin_nid == spec->port_map[i]) - return i + 1; + return i; } /* return -1 if pin number exceeds our expectation */ @@ -2684,9 +2684,9 @@ static int intel_port2pin(struct hda_codec *codec, int port) return port + intel_base_nid(codec) - 1; } - if (port < 1 || port > spec->port_num) + if (port < 0 || port >= spec->port_num) return 0; - return spec->port_map[port - 1]; + return spec->port_map[port]; } static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) @@ -2838,9 +2838,9 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) { /* * pin to port mapping table where the value indicate the pin number and - * the index indicate the port number with 1 base. + * the index indicate the port number. */ - static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb}; + static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); } @@ -2849,7 +2849,7 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) { /* * pin to port mapping table where the value indicate the pin number and - * the index indicate the port number with 1 base. + * the index indicate the port number. */ static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; From c8e3eb9be57f91751aeb9397118755390dbff9a4 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 15 Nov 2019 14:44:48 +0200 Subject: [PATCH 111/136] ALSA: hda: hdmi - remove redundant code comments Remove unnecessary comments related to pin mapping on Intel platforms. Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20191115124449.20512-3-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index dec90f931bae..d3768767625e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2655,7 +2655,7 @@ static int intel_pin2port(void *audio_ptr, int pin_nid) base_nid = intel_base_nid(codec); if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3)) return -1; - return pin_nid - base_nid + 1; /* intel port is 1-based */ + return pin_nid - base_nid + 1; } /* @@ -2667,7 +2667,6 @@ static int intel_pin2port(void *audio_ptr, int pin_nid) return i; } - /* return -1 if pin number exceeds our expectation */ codec_info(codec, "Can't find the HDMI/DP port for pin %d\n", pin_nid); return -1; } @@ -2680,7 +2679,6 @@ static int intel_port2pin(struct hda_codec *codec, int port) /* we assume only from port-B to port-D */ if (port < 1 || port > 3) return 0; - /* intel port is 1-based */ return port + intel_base_nid(codec) - 1; } @@ -2856,7 +2854,6 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); } - /* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { From ee906c6b9c6e7e4ad749f5cfdf65768455f4bf43 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 15 Nov 2019 14:44:49 +0200 Subject: [PATCH 112/136] ALSA: hda - remove forced polling workaround for CFL and CNL Remove the workarounds added in commit fa763f1b2858 ("ALSA: hda - Force polling mode on CNL for fixing codec communication") and commit a8d7bde23e71 ("ALSA: hda - Force polling mode on CFL for fixing codec communication"). The workarounds are no longer needed after the more generic change done in commit 2756d9143aa5 ("ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips"). This change applies to a larger set of hardware and covers CFL and CNL as well. Similar change was already done to SOF DSP HDA driver with no regressions detected. Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20191115124449.20512-4-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8981109e3565..e76a0bb6d3cf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -369,8 +369,6 @@ enum { ((pci)->device == 0x160c)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -#define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) -#define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -1763,10 +1761,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, if (!azx_snoop(chip)) azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_UC; - /* Workaround for a communication error on CFL (bko#199007) and CNL */ - if (IS_CFL(pci) || IS_CNL(pci)) - azx_bus(chip)->polling_mode = 1; - if (chip->driver_type == AZX_DRIVER_NVIDIA) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); chip->bus.needs_damn_long_delay = 1; From 80b917a8dd8f62ca00178d5a74d05d7471f21b47 Mon Sep 17 00:00:00 2001 From: Nikhil Mahale Date: Tue, 19 Nov 2019 14:17:07 +0530 Subject: [PATCH 113/136] ALSA: hda - Rename snd_hda_pin_sense to snd_hda_jack_pin_sense s/snd_hda_pin_sense/snd_hda_jack_pin_sense/g This aligns the snd_hda_pin_sense function name with the names of other functions in hda_jack.h. Signed-off-by: Nikhil Mahale Reviewed-by: Aaron Plattner Link: https://lore.kernel.org/r/20191119084710.29267-2-nmahale@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 8 ++++---- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 1fb7b06457ae..1ea42447278f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -191,14 +191,14 @@ void snd_hda_jack_set_dirty_all(struct hda_codec *codec) EXPORT_SYMBOL_GPL(snd_hda_jack_set_dirty_all); /** - * snd_hda_pin_sense - execute pin sense measurement + * snd_hda_jack_pin_sense - execute pin sense measurement * @codec: the CODEC to sense * @nid: the pin NID to sense * * Execute necessary pin sense measurement and return its Presence Detect, * Impedance, ELD Valid etc. status bits. */ -u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) +u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid) { struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); if (jack) { @@ -207,7 +207,7 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) } return read_pin_sense(codec, nid); } -EXPORT_SYMBOL_GPL(snd_hda_pin_sense); +EXPORT_SYMBOL_GPL(snd_hda_jack_pin_sense); /** * snd_hda_jack_detect_state - query pin Presence Detect status @@ -222,7 +222,7 @@ int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); if (jack && jack->phantom_jack) return HDA_JACK_PHANTOM; - else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + else if (snd_hda_jack_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) return HDA_JACK_PRESENT; else return HDA_JACK_NOT_PRESENT; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 22fe7ee43e82..cd9b47f51fab 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -65,7 +65,7 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); -u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); +u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid); /* the jack state returned from snd_hda_jack_detect_state() */ enum { diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d3768767625e..3b1b978201a9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1505,7 +1505,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, bool ret; bool do_repoll = false; - present = snd_hda_pin_sense(codec, pin_nid); + present = snd_hda_jack_pin_sense(codec, pin_nid); mutex_lock(&per_pin->lock); eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); From 5204a05d70d9354e6dd27219275c4b6725443dc4 Mon Sep 17 00:00:00 2001 From: Nikhil Mahale Date: Tue, 19 Nov 2019 14:17:08 +0530 Subject: [PATCH 114/136] ALSA: hda - Add DP-MST jack support This patch adds DP-MST jack support which will be used on NVIDIA platforms. Today, DP-MST audio is supported only if the codec has acomp support. This patch makes it possible to add DP-MST support for non-acomp codecs. For the codecs supporting DP-MST audio, each pin can contain several device entries. Each device entry is a virtual pin, described by pin_nid and dev_id in struct hdmi_spec_per_pin. For monitor hotplug event handling, non-acomp codecs enable and register jack-detection for every hdmi_spec_per_pin. This patch updates every relevant function in hda_jack.h and its implementation in hda_jack.c, to consider dev_id along with pin_nid. Changes to the HD Audio specification to support DP-MST audio are described in the Intel Document Change Notification (DCN) number HDA040-A. From HDA040-A, "For the case of multi stream capable Digital Display Pin Widget, [the Get Pin Sense verb] can be used to read a specific Device Entry state as reported in Get Device List Entry verb." This patch updates the read_pin_sense() function to take the dev_id as an argument and pass it as a parameter to the Get Pin Sense verb. Bits 15 through 20 from the Unsolicited Response for intrinsic events contain the index of the Device Entry that generated the event. This patch updates the Unsolicited Response event handlers to extract the device entry index from the response and pass it to snd_hda_jack_tbl_get_from_tag(). This patch updates snd_hda_jack_tbl_new() to take a dev_id argument and store it in the jack structure, and to make sure not to generate a different tag when called more than once for the same nid. Signed-off-by: Nikhil Mahale Link: https://lore.kernel.org/r/20191119084710.29267-3-nmahale@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 181 +++++++++++++++++++++++++------------ sound/pci/hda/hda_jack.h | 107 +++++++++++++++++++--- sound/pci/hda/patch_hdmi.c | 70 ++++++++------ 3 files changed, 261 insertions(+), 97 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 1ea42447278f..bf0255cb0515 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -43,7 +43,7 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_GPL(is_jack_detectable); /* execute pin sense measurement */ -static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) +static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id) { u32 pincap; u32 val; @@ -55,19 +55,57 @@ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) AC_VERB_SET_PIN_SENSE, 0); } val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); + AC_VERB_GET_PIN_SENSE, dev_id); if (codec->inv_jack_detect) val ^= AC_PINSENSE_PRESENCE; return val; } /** - * snd_hda_jack_tbl_get - query the jack-table entry for the given NID + * snd_hda_jack_tbl_get_mst - query the jack-table entry for the given NID * @codec: the HDA codec * @nid: pin NID to refer to + * @dev_id: pin device entry id */ struct hda_jack_tbl * -snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) +snd_hda_jack_tbl_get_mst(struct hda_codec *codec, hda_nid_t nid, int dev_id) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + if (!nid || !jack) + return NULL; + for (i = 0; i < codec->jacktbl.used; i++, jack++) + if (jack->nid == nid && jack->dev_id == dev_id) + return jack; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_mst); + +/** + * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag + * @codec: the HDA codec + * @tag: tag value to refer to + * @dev_id: pin device entry id + */ +struct hda_jack_tbl * +snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, + unsigned char tag, int dev_id) +{ + struct hda_jack_tbl *jack = codec->jacktbl.list; + int i; + + if (!tag || !jack) + return NULL; + for (i = 0; i < codec->jacktbl.used; i++, jack++) + if (jack->tag == tag && jack->dev_id == dev_id) + return jack; + return NULL; +} +EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag); + +static struct hda_jack_tbl * +any_jack_tbl_get_from_nid(struct hda_codec *codec, hda_nid_t nid) { struct hda_jack_tbl *jack = codec->jacktbl.list; int i; @@ -79,27 +117,6 @@ snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) return jack; return NULL; } -EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get); - -/** - * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag - * @codec: the HDA codec - * @tag: tag value to refer to - */ -struct hda_jack_tbl * -snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag) -{ - struct hda_jack_tbl *jack = codec->jacktbl.list; - int i; - - if (!tag || !jack) - return NULL; - for (i = 0; i < codec->jacktbl.used; i++, jack++) - if (jack->tag == tag) - return jack; - return NULL; -} -EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag); /** * snd_hda_jack_tbl_new - create a jack-table entry for the given NID @@ -107,17 +124,36 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag); * @nid: pin NID to assign */ static struct hda_jack_tbl * -snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) +snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid, int dev_id) { - struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + struct hda_jack_tbl *jack = + snd_hda_jack_tbl_get_mst(codec, nid, dev_id); + struct hda_jack_tbl *existing_nid_jack = + any_jack_tbl_get_from_nid(codec, nid); + + WARN_ON(dev_id != 0 && !codec->dp_mst); + if (jack) return jack; jack = snd_array_new(&codec->jacktbl); if (!jack) return NULL; jack->nid = nid; + jack->dev_id = dev_id; jack->jack_dirty = 1; - jack->tag = codec->jacktbl.used; + if (existing_nid_jack) { + jack->tag = existing_nid_jack->tag; + + /* + * Copy jack_detect from existing_nid_jack to avoid + * snd_hda_jack_detect_enable_callback_mst() making multiple + * SET_UNSOLICITED_ENABLE calls on the same pin. + */ + jack->jack_detect = existing_nid_jack->jack_detect; + } else { + jack->tag = codec->jacktbl.used; + } + return jack; } @@ -153,10 +189,12 @@ static void jack_detect_update(struct hda_codec *codec, if (jack->phantom_jack) jack->pin_sense = AC_PINSENSE_PRESENCE; else - jack->pin_sense = read_pin_sense(codec, jack->nid); + jack->pin_sense = read_pin_sense(codec, jack->nid, + jack->dev_id); /* A gating jack indicates the jack is invalid if gating is unplugged */ - if (jack->gating_jack && !snd_hda_jack_detect(codec, jack->gating_jack)) + if (jack->gating_jack && + !snd_hda_jack_detect_mst(codec, jack->gating_jack, jack->dev_id)) jack->pin_sense &= ~AC_PINSENSE_PRESENCE; jack->jack_dirty = 0; @@ -164,7 +202,8 @@ static void jack_detect_update(struct hda_codec *codec, /* If a jack is gated by this one update it. */ if (jack->gated_jack) { struct hda_jack_tbl *gated = - snd_hda_jack_tbl_get(codec, jack->gated_jack); + snd_hda_jack_tbl_get_mst(codec, jack->gated_jack, + jack->dev_id); if (gated) { gated->jack_dirty = 1; jack_detect_update(codec, gated); @@ -198,56 +237,62 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_set_dirty_all); * Execute necessary pin sense measurement and return its Presence Detect, * Impedance, ELD Valid etc. status bits. */ -u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid) +u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id) { - struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + struct hda_jack_tbl *jack = + snd_hda_jack_tbl_get_mst(codec, nid, dev_id); if (jack) { jack_detect_update(codec, jack); return jack->pin_sense; } - return read_pin_sense(codec, nid); + return read_pin_sense(codec, nid, dev_id); } EXPORT_SYMBOL_GPL(snd_hda_jack_pin_sense); /** - * snd_hda_jack_detect_state - query pin Presence Detect status + * snd_hda_jack_detect_state_mst - query pin Presence Detect status * @codec: the CODEC to sense * @nid: the pin NID to sense + * @dev_id: pin device entry id * * Query and return the pin's Presence Detect status, as either * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. */ -int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_state_mst(struct hda_codec *codec, + hda_nid_t nid, int dev_id) { - struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + struct hda_jack_tbl *jack = + snd_hda_jack_tbl_get_mst(codec, nid, dev_id); if (jack && jack->phantom_jack) return HDA_JACK_PHANTOM; - else if (snd_hda_jack_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + else if (snd_hda_jack_pin_sense(codec, nid, dev_id) & + AC_PINSENSE_PRESENCE) return HDA_JACK_PRESENT; else return HDA_JACK_NOT_PRESENT; } -EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state); +EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state_mst); /** - * snd_hda_jack_detect_enable - enable the jack-detection + * snd_hda_jack_detect_enable_mst - enable the jack-detection * @codec: the HDA codec * @nid: pin NID to enable * @func: callback function to register + * @dev_id: pin device entry id * * In the case of error, the return value will be a pointer embedded with * errno. Check and handle the return value appropriately with standard * macros such as @IS_ERR() and @PTR_ERR(). */ struct hda_jack_callback * -snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, - hda_jack_callback_fn func) +snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid, + int dev_id, hda_jack_callback_fn func) { struct hda_jack_tbl *jack; struct hda_jack_callback *callback = NULL; int err; - jack = snd_hda_jack_tbl_new(codec, nid); + jack = snd_hda_jack_tbl_new(codec, nid, dev_id); if (!jack) return ERR_PTR(-ENOMEM); if (func) { @@ -256,6 +301,7 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, return ERR_PTR(-ENOMEM); callback->func = func; callback->nid = jack->nid; + callback->dev_id = jack->dev_id; callback->next = jack->callback; jack->callback = callback; } @@ -272,19 +318,24 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, return ERR_PTR(err); return callback; } -EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback); +EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback_mst); /** * snd_hda_jack_detect_enable - Enable the jack detection on the given pin * @codec: the HDA codec * @nid: pin NID to enable jack detection + * @dev_id: pin device entry id * * Enable the jack detection with the default callback. Returns zero if * successful or a negative error code. */ -int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + int dev_id) { - return PTR_ERR_OR_ZERO(snd_hda_jack_detect_enable_callback(codec, nid, NULL)); + return PTR_ERR_OR_ZERO(snd_hda_jack_detect_enable_callback_mst(codec, + nid, + dev_id, + NULL)); } EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable); @@ -299,8 +350,11 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable); int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid) { - struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid); - struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid); + struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid, 0); + struct hda_jack_tbl *gating = + snd_hda_jack_tbl_new(codec, gating_nid, 0); + + WARN_ON(codec->dp_mst); if (!gated || !gating) return -EINVAL; @@ -376,9 +430,10 @@ static void hda_free_jack_priv(struct snd_jack *jack) } /** - * snd_hda_jack_add_kctl - Add a kctl for the given pin + * snd_hda_jack_add_kctl_mst - Add a kctl for the given pin * @codec: the HDA codec * @nid: pin NID to assign + * @dev_id : pin device entry id * @name: string name for the jack * @phantom_jack: flag to deal as a phantom jack * @type: jack type bits to be reported, 0 for guessing from pincfg @@ -387,15 +442,15 @@ static void hda_free_jack_priv(struct snd_jack *jack) * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. */ -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, bool phantom_jack, - int type, const struct hda_jack_keymap *keymap) +int snd_hda_jack_add_kctl_mst(struct hda_codec *codec, hda_nid_t nid, + int dev_id, const char *name, bool phantom_jack, + int type, const struct hda_jack_keymap *keymap) { struct hda_jack_tbl *jack; const struct hda_jack_keymap *map; int err, state, buttons; - jack = snd_hda_jack_tbl_new(codec, nid); + jack = snd_hda_jack_tbl_new(codec, nid, dev_id); if (!jack) return 0; if (jack->jack) @@ -425,12 +480,12 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, snd_jack_set_key(jack->jack, map->type, map->key); } - state = snd_hda_jack_detect(codec, nid); + state = snd_hda_jack_detect_mst(codec, nid, dev_id); snd_jack_report(jack->jack, state ? jack->type : 0); return 0; } -EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctl); +EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctl_mst); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, const struct auto_pin_cfg *cfg, @@ -441,6 +496,8 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int err; bool phantom_jack; + WARN_ON(codec->dp_mst); + if (!nid) return 0; def_conf = snd_hda_codec_get_pincfg(codec, nid); @@ -462,7 +519,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return err; if (!phantom_jack) - return snd_hda_jack_detect_enable(codec, nid); + return snd_hda_jack_detect_enable(codec, nid, 0); return 0; } @@ -540,7 +597,8 @@ static void call_jack_callback(struct hda_codec *codec, unsigned int res, } if (jack->gated_jack) { struct hda_jack_tbl *gated = - snd_hda_jack_tbl_get(codec, jack->gated_jack); + snd_hda_jack_tbl_get_mst(codec, jack->gated_jack, + jack->dev_id); if (gated) { for (cb = gated->callback; cb; cb = cb->next) { cb->jack = gated; @@ -561,7 +619,14 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) struct hda_jack_tbl *event; int tag = (res & AC_UNSOL_RES_TAG) >> AC_UNSOL_RES_TAG_SHIFT; - event = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (codec->dp_mst) { + int dev_entry = + (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; + + event = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry); + } else { + event = snd_hda_jack_tbl_get_from_tag(codec, tag, 0); + } if (!event) return; event->jack_dirty = 1; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index cd9b47f51fab..727b6d3ba454 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -19,6 +19,7 @@ typedef void (*hda_jack_callback_fn) (struct hda_codec *, struct hda_jack_callba struct hda_jack_callback { hda_nid_t nid; + int dev_id; hda_jack_callback_fn func; unsigned int private_data; /* arbitrary data */ unsigned int unsol_res; /* unsolicited event bits */ @@ -28,6 +29,7 @@ struct hda_jack_callback { struct hda_jack_tbl { hda_nid_t nid; + int dev_id; unsigned char tag; /* unsol event tag */ struct hda_jack_callback *callback; /* jack-detection stuff */ @@ -49,46 +51,129 @@ struct hda_jack_keymap { }; struct hda_jack_tbl * -snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid); +snd_hda_jack_tbl_get_mst(struct hda_codec *codec, hda_nid_t nid, int dev_id); + +/** + * snd_hda_jack_tbl_get - query the jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to refer to + */ +static inline struct hda_jack_tbl * +snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_tbl_get_mst(codec, nid, 0); +} + struct hda_jack_tbl * -snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag); +snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, + unsigned char tag, int dev_id); void snd_hda_jack_tbl_clear(struct hda_codec *codec); void snd_hda_jack_set_dirty_all(struct hda_codec *codec); -int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, + int dev_id); + struct hda_jack_callback * +snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid, + int dev_id, hda_jack_callback_fn cb); + +/** + * snd_hda_jack_detect_enable - enable the jack-detection + * @codec: the HDA codec + * @nid: pin NID to enable + * @func: callback function to register + * + * In the case of error, the return value will be a pointer embedded with + * errno. Check and handle the return value appropriately with standard + * macros such as @IS_ERR() and @PTR_ERR(). + */ +static inline struct hda_jack_callback * snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, - hda_jack_callback_fn cb); + hda_jack_callback_fn cb) +{ + return snd_hda_jack_detect_enable_callback_mst(codec, nid, 0, cb); +} int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); -u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid); +u32 snd_hda_jack_pin_sense(struct hda_codec *codec, hda_nid_t nid, int dev_id); /* the jack state returned from snd_hda_jack_detect_state() */ enum { HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM, }; -int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_jack_detect_state_mst(struct hda_codec *codec, hda_nid_t nid, + int dev_id); + +/** + * snd_hda_jack_detect_state - query pin Presence Detect status + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Query and return the pin's Presence Detect status, as either + * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. + */ +static inline int +snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_detect_state_mst(codec, nid, 0); +} + +/** + * snd_hda_jack_detect_mst - Detect the jack + * @codec: the HDA codec + * @nid: pin NID to check jack detection + * @dev_id: pin device entry id + */ +static inline bool +snd_hda_jack_detect_mst(struct hda_codec *codec, hda_nid_t nid, int dev_id) +{ + return snd_hda_jack_detect_state_mst(codec, nid, dev_id) != + HDA_JACK_NOT_PRESENT; +} /** * snd_hda_jack_detect - Detect the jack * @codec: the HDA codec * @nid: pin NID to check jack detection */ -static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +static inline bool +snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { - return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; + return snd_hda_jack_detect_mst(codec, nid, 0); } bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, bool phantom_jack, - int type, const struct hda_jack_keymap *keymap); +int snd_hda_jack_add_kctl_mst(struct hda_codec *codec, hda_nid_t nid, + int dev_id, const char *name, bool phantom_jack, + int type, const struct hda_jack_keymap *keymap); + +/** + * snd_hda_jack_add_kctl - Add a kctl for the given pin + * @codec: the HDA codec + * @nid: pin NID to assign + * @name: string name for the jack + * @phantom_jack: flag to deal as a phantom jack + * @type: jack type bits to be reported, 0 for guessing from pincfg + * @keymap: optional jack / key mapping + * + * This assigns a jack-detection kctl to the given pin. The kcontrol + * will have the given name and index. + */ +static inline int +snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, + const char *name, bool phantom_jack, + int type, const struct hda_jack_keymap *keymap) +{ + return snd_hda_jack_add_kctl_mst(codec, nid, 0, + name, phantom_jack, type, keymap); +} + int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3b1b978201a9..37b4345afdde 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -758,34 +758,32 @@ static void jack_callback(struct hda_codec *codec, if (codec_has_acomp(codec)) return; - /* hda_jack don't support DP MST */ - check_presence_and_report(codec, jack->nid, 0); + check_presence_and_report(codec, jack->nid, jack->dev_id); } static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { int tag = res >> AC_UNSOL_RES_TAG_SHIFT; struct hda_jack_tbl *jack; - int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; - /* - * assume DP MST uses dyn_pcm_assign and acomp and - * never comes here - * if DP MST supports unsol event, below code need - * consider dev_entry - */ - jack = snd_hda_jack_tbl_get_from_tag(codec, tag); + if (codec->dp_mst) { + int dev_entry = + (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; + + jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry); + } else { + jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0); + } if (!jack) return; jack->jack_dirty = 1; codec_dbg(codec, "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA), + codec->addr, jack->nid, jack->dev_id, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); - /* hda_jack don't support DP MST */ - check_presence_and_report(codec, jack->nid, 0); + check_presence_and_report(codec, jack->nid, jack->dev_id); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -815,11 +813,21 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) { int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + struct hda_jack_tbl *jack; if (codec_has_acomp(codec)) return; - if (!snd_hda_jack_tbl_get_from_tag(codec, tag)) { + if (codec->dp_mst) { + int dev_entry = + (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; + + jack = snd_hda_jack_tbl_get_from_tag(codec, tag, dev_entry); + } else { + jack = snd_hda_jack_tbl_get_from_tag(codec, tag, 0); + } + + if (!jack) { codec_dbg(codec, "Unexpected HDMI event tag 0x%x\n", tag); return; } @@ -1505,7 +1513,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, bool ret; bool do_repoll = false; - present = snd_hda_jack_pin_sense(codec, pin_nid); + present = snd_hda_jack_pin_sense(codec, pin_nid, per_pin->dev_id); mutex_lock(&per_pin->lock); eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); @@ -1538,7 +1546,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, ret = !repoll || !eld->monitor_present || eld->eld_valid; - jack = snd_hda_jack_tbl_get(codec, pin_nid); + jack = snd_hda_jack_tbl_get_mst(codec, pin_nid, per_pin->dev_id); if (jack) { jack->block_report = !ret; jack->pin_sense = (eld->monitor_present && eld->eld_valid) ? @@ -1569,7 +1577,8 @@ static struct snd_jack *pin_idx_to_jack(struct hda_codec *codec, * DP MST will use dyn_pcm_assign, * so DP MST will never come here */ - jack_tbl = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + jack_tbl = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, + per_pin->dev_id); if (jack_tbl) jack = jack_tbl->jack; } @@ -1650,7 +1659,8 @@ static void hdmi_repoll_eld(struct work_struct *work) struct hdmi_spec *spec = codec->spec; struct hda_jack_tbl *jack; - jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, + per_pin->dev_id); if (jack) jack->jack_dirty = 1; @@ -2151,11 +2161,13 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pcm_idx) if (phantom_jack) strncat(hdmi_str, " Phantom", sizeof(hdmi_str) - strlen(hdmi_str) - 1); - ret = snd_hda_jack_add_kctl(codec, per_pin->pin_nid, hdmi_str, - phantom_jack, 0, NULL); + ret = snd_hda_jack_add_kctl_mst(codec, per_pin->pin_nid, + per_pin->dev_id, hdmi_str, phantom_jack, + 0, NULL); if (ret < 0) return ret; - jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + jack = snd_hda_jack_tbl_get_mst(codec, per_pin->pin_nid, + per_pin->dev_id); if (jack == NULL) return 0; /* assign jack->jack to pcm_rec[].jack to @@ -2264,10 +2276,11 @@ static int generic_hdmi_init(struct hda_codec *codec) if (codec_has_acomp(codec)) continue; if (spec->use_jack_detect) - snd_hda_jack_detect_enable(codec, pin_nid); + snd_hda_jack_detect_enable(codec, pin_nid, dev_id); else - snd_hda_jack_detect_enable_callback(codec, pin_nid, - jack_callback); + snd_hda_jack_detect_enable_callback_mst(codec, pin_nid, + dev_id, + jack_callback); } mutex_unlock(&spec->bind_lock); return 0; @@ -2417,11 +2430,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) /* turn on / off the unsol event jack detection dynamically */ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid, - bool use_acomp) + int dev_id, bool use_acomp) { struct hda_jack_tbl *tbl; - tbl = snd_hda_jack_tbl_get(codec, nid); + tbl = snd_hda_jack_tbl_get_mst(codec, nid, dev_id); if (tbl) { /* clear unsol even if component notifier is used, or re-enable * if notifier is cleared @@ -2434,7 +2447,7 @@ static void reprogram_jack_detect(struct hda_codec *codec, hda_nid_t nid, * at need (i.e. only when notifier is cleared) */ if (!use_acomp) - snd_hda_jack_detect_enable(codec, nid); + snd_hda_jack_detect_enable(codec, nid, dev_id); } } @@ -2454,6 +2467,7 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp, for (i = 0; i < spec->num_pins; i++) reprogram_jack_detect(spec->codec, get_pin(spec, i)->pin_nid, + get_pin(spec, i)->dev_id, use_acomp); } mutex_unlock(&spec->bind_lock); @@ -2959,7 +2973,7 @@ static int simple_playback_init(struct hda_codec *codec) if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - snd_hda_jack_detect_enable(codec, pin); + snd_hda_jack_detect_enable(codec, pin, per_pin->dev_id); return 0; } From 9c32fea836928d7a25a83b337f268e533cfc5c3d Mon Sep 17 00:00:00 2001 From: Nikhil Mahale Date: Tue, 19 Nov 2019 14:17:09 +0530 Subject: [PATCH 115/136] ALSA: hda - Add DP-MST support for non-acomp codecs This patch make it possible for non-acomp codecs to set dyn_pcm_assign/dp_mst and get DP-MST audio support. Document change notification HDA040-A for the Intel High Definition Audio 1.0a specification introduces a Device Select verb for Digital Display Pin Widgets that are multi-stream capable. This verb selects a Device Entry that is used by subsequent Pin Widget verbs. Once the Device Entry is selected, all subsequent Pin Widget verbs controlling the sink device will be directed to the selected Device Entry until the Device Select verb is updated with a new value. These Pin Widget verbs include: * Connection Select * Get Connection List Entry * Amplifier Gain/Mute * Power State * Pin Widget Control * ELD Data * DIP-Size * DIP-Index * DIP-Data * DIP-XmitCtrl * Content Protection Control * ASP Channel Mapping This patch adds calls to snd_hda_set_dev_select() to direct each of these Pin Widget control verbs to the correct Device Entry. snd_hda_get_connections() does not return per-device connection list, therefore make use snd_hda_get_raw_connections() instead of snd_hda_get_connections(). Signed-off-by: Nikhil Mahale Reviewed-by: Aaron Plattner Link: https://lore.kernel.org/r/20191119084710.29267-4-nmahale@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 100 +++++++++++++++++++++++++------------ 1 file changed, 67 insertions(+), 33 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 37b4345afdde..0a3045d49297 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -80,16 +80,19 @@ struct hdmi_spec_per_pin { /* operations used by generic code that can be overridden by patches */ struct hdmi_ops { int (*pin_get_eld)(struct hda_codec *codec, hda_nid_t pin_nid, - unsigned char *buf, int *eld_size); + int dev_id, unsigned char *buf, int *eld_size); void (*pin_setup_infoframe)(struct hda_codec *codec, hda_nid_t pin_nid, + int dev_id, int ca, int active_channels, int conn_type); /* enable/disable HBR (HD passthrough) */ - int (*pin_hbr_setup)(struct hda_codec *codec, hda_nid_t pin_nid, bool hbr); + int (*pin_hbr_setup)(struct hda_codec *codec, hda_nid_t pin_nid, + int dev_id, bool hbr); int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid, - hda_nid_t pin_nid, u32 stream_tag, int format); + hda_nid_t pin_nid, int dev_id, u32 stream_tag, + int format); void (*pin_cvt_fixup)(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin, @@ -636,8 +639,16 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } +static int hdmi_pin_get_eld(struct hda_codec *codec, hda_nid_t nid, + int dev_id, unsigned char *buf, int *eld_size) +{ + snd_hda_set_dev_select(codec, nid, dev_id); + + return snd_hdmi_get_eld(codec, nid, buf, eld_size); +} + static void hdmi_pin_setup_infoframe(struct hda_codec *codec, - hda_nid_t pin_nid, + hda_nid_t pin_nid, int dev_id, int ca, int active_channels, int conn_type) { @@ -667,6 +678,8 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, return; } + snd_hda_set_dev_select(codec, pin_nid, dev_id); + /* * sizeof(ai) is used instead of sizeof(*hdmi_ai) or * sizeof(*dp_ai) to avoid partial match/update problems when @@ -692,6 +705,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, struct hdmi_spec *spec = codec->spec; struct hdac_chmap *chmap = &spec->chmap; hda_nid_t pin_nid = per_pin->pin_nid; + int dev_id = per_pin->dev_id; int channels = per_pin->channels; int active_channels; struct hdmi_eld *eld; @@ -700,6 +714,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + snd_hda_set_dev_select(codec, pin_nid, dev_id); + /* some HW (e.g. HSW+) needs reprogramming the amp at each time */ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, @@ -725,8 +741,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, pin_nid, non_pcm, ca, channels, per_pin->chmap, per_pin->chmap_set); - spec->ops.pin_setup_infoframe(codec, pin_nid, ca, active_channels, - eld->info.conn_type); + spec->ops.pin_setup_infoframe(codec, pin_nid, dev_id, + ca, active_channels, eld->info.conn_type); per_pin->non_pcm = non_pcm; } @@ -868,11 +884,12 @@ static void haswell_verify_D0(struct hda_codec *codec, ((format & AC_FMT_TYPE_NON_PCM) && (format & AC_FMT_CHAN_MASK) == 7) static int hdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, - bool hbr) + int dev_id, bool hbr) { int pinctl, new_pinctl; if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { + snd_hda_set_dev_select(codec, pin_nid, dev_id); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); @@ -902,13 +919,15 @@ static int hdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, } static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, - hda_nid_t pin_nid, u32 stream_tag, int format) + hda_nid_t pin_nid, int dev_id, + u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; unsigned int param; int err; - err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); + err = spec->ops.pin_hbr_setup(codec, pin_nid, dev_id, + is_hbr_format(format)); if (err) { codec_dbg(codec, "hdmi_setup_stream: HBR is not supported\n"); @@ -1282,6 +1301,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; + int dev_id = per_pin->dev_id; if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { codec_warn(codec, @@ -1290,10 +1310,12 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return -EINVAL; } + snd_hda_set_dev_select(codec, pin_nid, dev_id); + /* all the device entries on the same pin have the same conn list */ - per_pin->num_mux_nids = snd_hda_get_connections(codec, pin_nid, - per_pin->mux_nids, - HDA_MAX_CONNECTIONS); + per_pin->num_mux_nids = + snd_hda_get_raw_connections(codec, pin_nid, per_pin->mux_nids, + HDA_MAX_CONNECTIONS); return 0; } @@ -1501,6 +1523,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; hda_nid_t pin_nid = per_pin->pin_nid; + int dev_id = per_pin->dev_id; /* * Always execute a GetPinSense verb here, even when called from * hdmi_intrinsic_event; for some NVIDIA HW, the unsolicited @@ -1513,7 +1536,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, bool ret; bool do_repoll = false; - present = snd_hda_jack_pin_sense(codec, pin_nid, per_pin->dev_id); + present = snd_hda_jack_pin_sense(codec, pin_nid, dev_id); mutex_lock(&per_pin->lock); eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); @@ -1527,8 +1550,8 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { - if (spec->ops.pin_get_eld(codec, pin_nid, eld->eld_buffer, - &eld->eld_size) < 0) + if (spec->ops.pin_get_eld(codec, pin_nid, dev_id, + eld->eld_buffer, &eld->eld_size) < 0) eld->eld_valid = false; else { if (snd_hdmi_parse_eld(codec, &eld->info, eld->eld_buffer, @@ -1865,7 +1888,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx; struct hdmi_spec_per_pin *per_pin; - hda_nid_t pin_nid; struct snd_pcm_runtime *runtime = substream->runtime; bool non_pcm; int pinctl, stripe; @@ -1889,7 +1911,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, goto unlock; } per_pin = get_pin(spec, pin_idx); - pin_nid = per_pin->pin_nid; /* Verify pin:cvt selections to avoid silent audio after S3. * After S3, the audio driver restores pin:cvt selections @@ -1904,8 +1925,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->core, pin_nid, per_pin->dev_id, - runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, + per_pin->dev_id, runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -1923,16 +1944,18 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); mutex_unlock(&per_pin->lock); if (spec->dyn_pin_out) { - pinctl = snd_hda_codec_read(codec, pin_nid, 0, + snd_hda_set_dev_select(codec, per_pin->pin_nid, + per_pin->dev_id); + pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, pin_nid, 0, + snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); } /* snd_hda_set_dev_select() has been called before */ - err = spec->ops.setup_stream(codec, cvt_nid, pin_nid, - stream_tag, format); + err = spec->ops.setup_stream(codec, cvt_nid, per_pin->pin_nid, + per_pin->dev_id, stream_tag, format); unlock: mutex_unlock(&spec->pcm_lock); return err; @@ -1984,6 +2007,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_pin = get_pin(spec, pin_idx); if (spec->dyn_pin_out) { + snd_hda_set_dev_select(codec, per_pin->pin_nid, + per_pin->dev_id); pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); snd_hda_codec_write(codec, per_pin->pin_nid, 0, @@ -2370,7 +2395,7 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { }; static const struct hdmi_ops generic_standard_hdmi_ops = { - .pin_get_eld = snd_hdmi_get_eld, + .pin_get_eld = hdmi_pin_get_eld, .pin_setup_infoframe = hdmi_pin_setup_infoframe, .pin_hbr_setup = hdmi_pin_hbr_setup, .setup_stream = hdmi_setup_stream, @@ -2568,7 +2593,8 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t conns[4]; int nconns; - nconns = snd_hda_get_connections(codec, nid, conns, ARRAY_SIZE(conns)); + nconns = snd_hda_get_raw_connections(codec, nid, conns, + ARRAY_SIZE(conns)); if (nconns == spec->num_cvts && !memcmp(conns, spec->cvt_nids, spec->num_cvts * sizeof(hda_nid_t))) return; @@ -2744,10 +2770,12 @@ static void register_i915_notifier(struct hda_codec *codec) /* setup_stream ops override for HSW+ */ static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, - hda_nid_t pin_nid, u32 stream_tag, int format) + hda_nid_t pin_nid, int dev_id, u32 stream_tag, + int format) { haswell_verify_D0(codec, cvt_nid, pin_nid); - return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, dev_id, + stream_tag, format); } /* pin_cvt_fixup ops override for HSW+ and VLV+ */ @@ -3713,16 +3741,19 @@ static int patch_tegra_hdmi(struct hda_codec *codec) #define ATI_HBR_ENABLE 0x10 static int atihdmi_pin_get_eld(struct hda_codec *codec, hda_nid_t nid, - unsigned char *buf, int *eld_size) + int dev_id, unsigned char *buf, int *eld_size) { + WARN_ON(dev_id != 0); /* call hda_eld.c ATI/AMD-specific function */ return snd_hdmi_get_eld_ati(codec, nid, buf, eld_size, is_amdhdmi_rev3_or_later(codec)); } -static void atihdmi_pin_setup_infoframe(struct hda_codec *codec, hda_nid_t pin_nid, int ca, +static void atihdmi_pin_setup_infoframe(struct hda_codec *codec, + hda_nid_t pin_nid, int dev_id, int ca, int active_channels, int conn_type) { + WARN_ON(dev_id != 0); snd_hda_codec_write(codec, pin_nid, 0, ATI_VERB_SET_CHANNEL_ALLOCATION, ca); } @@ -3913,10 +3944,12 @@ static void atihdmi_paired_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, } static int atihdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, - bool hbr) + int dev_id, bool hbr) { int hbr_ctl, hbr_ctl_new; + WARN_ON(dev_id != 0); + hbr_ctl = snd_hda_codec_read(codec, pin_nid, 0, ATI_VERB_GET_HBR_CONTROL, 0); if (hbr_ctl >= 0 && (hbr_ctl & ATI_HBR_CAPABLE)) { if (hbr) @@ -3942,9 +3975,9 @@ static int atihdmi_pin_hbr_setup(struct hda_codec *codec, hda_nid_t pin_nid, } static int atihdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, - hda_nid_t pin_nid, u32 stream_tag, int format) + hda_nid_t pin_nid, int dev_id, + u32 stream_tag, int format) { - if (is_amdhdmi_rev3_or_later(codec)) { int ramp_rate = 180; /* default as per AMD spec */ /* disable ramp-up/down for non-pcm as per AMD spec */ @@ -3954,7 +3987,8 @@ static int atihdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, snd_hda_codec_write(codec, cvt_nid, 0, ATI_VERB_SET_RAMP_RATE, ramp_rate); } - return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, dev_id, + stream_tag, format); } From 5398e94fb753d022301825ebfa5f7cf8a660d8eb Mon Sep 17 00:00:00 2001 From: Nikhil Mahale Date: Tue, 19 Nov 2019 14:17:10 +0530 Subject: [PATCH 116/136] ALSA: hda - Add DP-MST support for NVIDIA codecs This patch adds DP-MST support for GK104+ NVIDIA codecs. GK104+ NVIDIA codecs support DP-MST audio. These codecs have 4 output converters and 4 pin widgets, with 4 device entries per pin widget for a total of 16 device entries. This patch moves the existing patch_nvhdmi() definition to patch_nvhdmi_legacy(), used by pre-GK104 NVIDIA codecs. Redefine patch_nvhdmi() to enable DP-MST support by setting codec->dp_mst and spec->dyn_pcm_assign. Introduce fresh logic for dynamic pcm assignment, making sure that new pcm assignments are compatible with the legacy static per_pin-pmc assignment that existed in the days before DP-MST. Signed-off-by: Nikhil Mahale Reviewed-by: Aaron Plattner Link: https://lore.kernel.org/r/20191119084710.29267-5-nmahale@nvidia.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 95 +++++++++++++++++++++++++++++--------- 1 file changed, 73 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0a3045d49297..55d20e40a195 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1321,15 +1321,32 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) } static int hdmi_find_pcm_slot(struct hdmi_spec *spec, - struct hdmi_spec_per_pin *per_pin) + struct hdmi_spec_per_pin *per_pin) { int i; - /* try the prefer PCM */ - if (!test_bit(per_pin->pin_nid_idx, &spec->pcm_bitmap)) + /* + * generic_hdmi_build_pcms() allocates (num_nids + dev_num - 1) + * number of pcms. + * + * The per_pin of pin_nid_idx=n and dev_id=m prefers to get pcm-n + * if m==0. This guarantees that dynamic pcm assignments are compatible + * with the legacy static per_pin-pmc assignment that existed in the + * days before DP-MST. + * + * per_pin of m!=0 prefers to get pcm=(num_nids + (m - 1)). + */ + if (per_pin->dev_id == 0 && + !test_bit(per_pin->pin_nid_idx, &spec->pcm_bitmap)) return per_pin->pin_nid_idx; - /* have a second try; check the "reserved area" over num_pins */ + if (per_pin->dev_id != 0 && + !(test_bit(spec->num_nids + (per_pin->dev_id - 1), + &spec->pcm_bitmap))) { + return spec->num_nids + (per_pin->dev_id - 1); + } + + /* have a second try; check the area over num_nids */ for (i = spec->num_nids; i < spec->pcm_used; i++) { if (!test_bit(i, &spec->pcm_bitmap)) return i; @@ -3510,6 +3527,40 @@ static int patch_nvhdmi(struct hda_codec *codec) struct hdmi_spec *spec; int err; + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + codec->dp_mst = true; + + spec = codec->spec; + spec->dyn_pcm_assign = true; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; + } + + generic_hdmi_init_per_pins(codec); + + spec->dyn_pin_out = true; + + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + + codec->link_down_at_suspend = 1; + + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); + + return 0; +} + +static int patch_nvhdmi_legacy(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + err = patch_generic_hdmi(codec); if (err) return err; @@ -4118,25 +4169,25 @@ HDA_CODEC_ENTRY(0x10de0004, "GPU 04 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0005, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0006, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x), HDA_CODEC_ENTRY(0x10de0007, "MCP79/7A HDMI", patch_nvhdmi_8ch_7x), -HDA_CODEC_ENTRY(0x10de0008, "GPU 08 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0009, "GPU 09 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de000d, "GPU 0d HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0010, "GPU 10 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0011, "GPU 11 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0012, "GPU 12 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0013, "GPU 13 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0014, "GPU 14 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0015, "GPU 15 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0016, "GPU 16 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0008, "GPU 08 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0009, "GPU 09 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de000d, "GPU 0d HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0010, "GPU 10 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0011, "GPU 11 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0012, "GPU 12 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0013, "GPU 13 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0014, "GPU 14 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0015, "GPU 15 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0016, "GPU 16 HDMI/DP", patch_nvhdmi_legacy), /* 17 is known to be absent */ -HDA_CODEC_ENTRY(0x10de0018, "GPU 18 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de0019, "GPU 19 HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de001a, "GPU 1a HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de001b, "GPU 1b HDMI/DP", patch_nvhdmi), -HDA_CODEC_ENTRY(0x10de001c, "GPU 1c HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0018, "GPU 18 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de0019, "GPU 19 HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de001a, "GPU 1a HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de001b, "GPU 1b HDMI/DP", patch_nvhdmi_legacy), +HDA_CODEC_ENTRY(0x10de001c, "GPU 1c HDMI/DP", patch_nvhdmi_legacy), HDA_CODEC_ENTRY(0x10de0020, "Tegra30 HDMI", patch_tegra_hdmi), HDA_CODEC_ENTRY(0x10de0022, "Tegra114 HDMI", patch_tegra_hdmi), HDA_CODEC_ENTRY(0x10de0028, "Tegra124 HDMI", patch_tegra_hdmi), From e190de6941db14813032af87873f5550ad5764fe Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Wed, 20 Nov 2019 16:20:35 +0800 Subject: [PATCH 117/136] ALSA: hda - Add mute led support for HP ProBook 645 G4 Mic mute led does not work on HP ProBook 645 G4. We can use CXT_FIXUP_MUTE_LED_GPIO fixup to support it. Signed-off-by: Kai-Heng Feng Cc: Link: https://lore.kernel.org/r/20191120082035.18937-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 968d3caab6ac..90aa0f400a57 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -910,6 +910,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), From 97dda3da20732df8010090dd6d749b9d5b86bffe Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Wed, 20 Nov 2019 11:49:49 -0600 Subject: [PATCH 118/136] ALSA: aloop: Describe units of variables Describe the unit of the variables used to calculate the hw pointer depending on jiffies ticks. Signed-off-by: Timo Wischer Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-2-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 54f8b17476a1..573b06cf7cf5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -102,8 +102,10 @@ struct loopback_pcm { /* flags */ unsigned int period_update_pending :1; /* timer stuff */ - unsigned int irq_pos; /* fractional IRQ position */ - unsigned int period_size_frac; + unsigned int irq_pos; /* fractional IRQ position in jiffies + * ticks + */ + unsigned int period_size_frac; /* period size in jiffies ticks */ unsigned int last_drift; unsigned long last_jiffies; struct timer_list timer; From 09419f1ace213284823ad835d28b239c33daeb71 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Wed, 20 Nov 2019 11:49:50 -0600 Subject: [PATCH 119/136] ALSA: aloop: Support return of error code for timer start and stop This is required for additional timer implementations which could detect errors and want to throw them. Signed-off-by: Timo Wischer Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-3-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 30 +++++++++++++++++++----------- 1 file changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 573b06cf7cf5..7919006f70a5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -155,7 +155,7 @@ static inline unsigned int get_rate_shift(struct loopback_pcm *dpcm) } /* call in cable->lock */ -static void loopback_timer_start(struct loopback_pcm *dpcm) +static int loopback_timer_start(struct loopback_pcm *dpcm) { unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); @@ -171,18 +171,24 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) tick = dpcm->period_size_frac - dpcm->irq_pos; tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; mod_timer(&dpcm->timer, jiffies + tick); + + return 0; } /* call in cable->lock */ -static inline void loopback_timer_stop(struct loopback_pcm *dpcm) +static inline int loopback_timer_stop(struct loopback_pcm *dpcm) { del_timer(&dpcm->timer); dpcm->timer.expires = 0; + + return 0; } -static inline void loopback_timer_stop_sync(struct loopback_pcm *dpcm) +static inline int loopback_timer_stop_sync(struct loopback_pcm *dpcm) { del_timer_sync(&dpcm->timer); + + return 0; } #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) @@ -251,7 +257,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; - int err, stream = 1 << substream->stream; + int err = 0, stream = 1 << substream->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -264,7 +270,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&cable->lock); cable->running |= stream; cable->pause &= ~stream; - loopback_timer_start(dpcm); + err = loopback_timer_start(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -273,7 +279,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&cable->lock); cable->running &= ~stream; cable->pause &= ~stream; - loopback_timer_stop(dpcm); + err = loopback_timer_stop(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -282,7 +288,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&cable->lock); cable->pause |= stream; - loopback_timer_stop(dpcm); + err = loopback_timer_stop(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -292,7 +298,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&cable->lock); dpcm->last_jiffies = jiffies; cable->pause &= ~stream; - loopback_timer_start(dpcm); + err = loopback_timer_start(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -300,7 +306,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) default: return -EINVAL; } - return 0; + return err; } static void params_change(struct snd_pcm_substream *substream) @@ -321,9 +327,11 @@ static int loopback_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; - int bps, salign; + int err, bps, salign; - loopback_timer_stop_sync(dpcm); + err = loopback_timer_stop_sync(dpcm); + if (err < 0) + return err; salign = (snd_pcm_format_physical_width(runtime->format) * runtime->channels) / 8; From 133f37593eb6db43158bafd52937974acccd1d29 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Wed, 20 Nov 2019 11:49:51 -0600 Subject: [PATCH 120/136] ALSA: aloop: Use callback functions for timer specific implementations This commit only refactors the implementation. It does not change the behaviour. It is required to support other timers (e.g sound timer). Signed-off-by: Timo Wischer Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-4-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 113 +++++++++++++++++++++++++++++++++++------- 1 file changed, 94 insertions(+), 19 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 7919006f70a5..3bfd7c32803c 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -55,8 +55,39 @@ MODULE_PARM_DESC(pcm_notify, "Break capture when PCM format/rate/channels change #define NO_PITCH 100000 +struct loopback_cable; struct loopback_pcm; +struct loopback_ops { + /* optional + * call in loopback->cable_lock + */ + int (*open)(struct loopback_pcm *dpcm); + /* required + * call in cable->lock + */ + int (*start)(struct loopback_pcm *dpcm); + /* required + * call in cable->lock + */ + int (*stop)(struct loopback_pcm *dpcm); + /* optional */ + int (*stop_sync)(struct loopback_pcm *dpcm); + /* optional */ + int (*close_substream)(struct loopback_pcm *dpcm); + /* optional + * call in loopback->cable_lock + */ + int (*close_cable)(struct loopback_pcm *dpcm); + /* optional + * call in cable->lock + */ + unsigned int (*pos_update)(struct loopback_cable *cable); + /* optional */ + void (*dpcm_info)(struct loopback_pcm *dpcm, + struct snd_info_buffer *buffer); +}; + struct loopback_cable { spinlock_t lock; struct loopback_pcm *streams[2]; @@ -65,6 +96,8 @@ struct loopback_cable { unsigned int valid; unsigned int running; unsigned int pause; + /* timer specific */ + struct loopback_ops *ops; }; struct loopback_setup { @@ -270,7 +303,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&cable->lock); cable->running |= stream; cable->pause &= ~stream; - err = loopback_timer_start(dpcm); + err = cable->ops->start(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -279,7 +312,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&cable->lock); cable->running &= ~stream; cable->pause &= ~stream; - err = loopback_timer_stop(dpcm); + err = cable->ops->stop(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -288,7 +321,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&cable->lock); cable->pause |= stream; - err = loopback_timer_stop(dpcm); + err = cable->ops->stop(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -298,7 +331,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&cable->lock); dpcm->last_jiffies = jiffies; cable->pause &= ~stream; - err = loopback_timer_start(dpcm); + err = cable->ops->start(dpcm); spin_unlock(&cable->lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -329,9 +362,11 @@ static int loopback_prepare(struct snd_pcm_substream *substream) struct loopback_cable *cable = dpcm->cable; int err, bps, salign; - err = loopback_timer_stop_sync(dpcm); - if (err < 0) - return err; + if (cable->ops->stop_sync) { + err = cable->ops->stop_sync(dpcm); + if (err < 0) + return err; + } salign = (snd_pcm_format_physical_width(runtime->format) * runtime->channels) / 8; @@ -539,6 +574,18 @@ static void loopback_timer_function(struct timer_list *t) spin_unlock_irqrestore(&dpcm->cable->lock, flags); } +static void loopback_jiffies_timer_dpcm_info(struct loopback_pcm *dpcm, + struct snd_info_buffer *buffer) +{ + snd_iprintf(buffer, " update_pending:\t%u\n", + dpcm->period_update_pending); + snd_iprintf(buffer, " irq_pos:\t\t%u\n", dpcm->irq_pos); + snd_iprintf(buffer, " period_frac:\t%u\n", dpcm->period_size_frac); + snd_iprintf(buffer, " last_jiffies:\t%lu (%lu)\n", + dpcm->last_jiffies, jiffies); + snd_iprintf(buffer, " timer_expires:\t%lu\n", dpcm->timer.expires); +} + static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -546,7 +593,8 @@ static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t pos; spin_lock(&dpcm->cable->lock); - loopback_pos_update(dpcm->cable); + if (dpcm->cable->ops->pos_update) + dpcm->cable->ops->pos_update(dpcm->cable); pos = dpcm->buf_pos; spin_unlock(&dpcm->cable->lock); return bytes_to_frames(runtime, pos); @@ -671,12 +719,33 @@ static void free_cable(struct snd_pcm_substream *substream) cable->streams[substream->stream] = NULL; spin_unlock_irq(&cable->lock); } else { + struct loopback_pcm *dpcm = substream->runtime->private_data; + + if (cable->ops && cable->ops->close_cable && dpcm) + cable->ops->close_cable(dpcm); /* free the cable */ loopback->cables[substream->number][dev] = NULL; kfree(cable); } } +static int loopback_jiffies_timer_open(struct loopback_pcm *dpcm) +{ + timer_setup(&dpcm->timer, loopback_timer_function, 0); + + return 0; +} + +static struct loopback_ops loopback_jiffies_timer_ops = { + .open = loopback_jiffies_timer_open, + .start = loopback_timer_start, + .stop = loopback_timer_stop, + .stop_sync = loopback_timer_stop_sync, + .close_substream = loopback_timer_stop_sync, + .pos_update = loopback_pos_update, + .dpcm_info = loopback_jiffies_timer_dpcm_info, +}; + static int loopback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -694,7 +763,6 @@ static int loopback_open(struct snd_pcm_substream *substream) } dpcm->loopback = loopback; dpcm->substream = substream; - timer_setup(&dpcm->timer, loopback_timer_function, 0); cable = loopback->cables[substream->number][dev]; if (!cable) { @@ -705,9 +773,17 @@ static int loopback_open(struct snd_pcm_substream *substream) } spin_lock_init(&cable->lock); cable->hw = loopback_pcm_hardware; + cable->ops = &loopback_jiffies_timer_ops; loopback->cables[substream->number][dev] = cable; } dpcm->cable = cable; + runtime->private_data = dpcm; + + if (cable->ops->open) { + err = cable->ops->open(dpcm); + if (err < 0) + goto unlock; + } snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -733,7 +809,9 @@ static int loopback_open(struct snd_pcm_substream *substream) if (err < 0) goto unlock; - runtime->private_data = dpcm; + /* loopback_runtime_free() has not to be called if kfree(dpcm) was + * already called here. Otherwise it will end up with a double free. + */ runtime->private_free = loopback_runtime_free; if (get_notify(dpcm)) runtime->hw = loopback_pcm_hardware; @@ -757,12 +835,14 @@ static int loopback_close(struct snd_pcm_substream *substream) { struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm = substream->runtime->private_data; + int err = 0; - loopback_timer_stop_sync(dpcm); + if (dpcm->cable->ops->close_substream) + err = dpcm->cable->ops->close_substream(dpcm); mutex_lock(&loopback->cable_lock); free_cable(substream); mutex_unlock(&loopback->cable_lock); - return 0; + return err; } static const struct snd_pcm_ops loopback_pcm_ops = { @@ -1086,13 +1166,8 @@ static void print_dpcm_info(struct snd_info_buffer *buffer, snd_iprintf(buffer, " bytes_per_sec:\t%u\n", dpcm->pcm_bps); snd_iprintf(buffer, " sample_align:\t%u\n", dpcm->pcm_salign); snd_iprintf(buffer, " rate_shift:\t\t%u\n", dpcm->pcm_rate_shift); - snd_iprintf(buffer, " update_pending:\t%u\n", - dpcm->period_update_pending); - snd_iprintf(buffer, " irq_pos:\t\t%u\n", dpcm->irq_pos); - snd_iprintf(buffer, " period_frac:\t%u\n", dpcm->period_size_frac); - snd_iprintf(buffer, " last_jiffies:\t%lu (%lu)\n", - dpcm->last_jiffies, jiffies); - snd_iprintf(buffer, " timer_expires:\t%lu\n", dpcm->timer.expires); + if (dpcm->cable->ops->dpcm_info) + dpcm->cable->ops->dpcm_info(dpcm, buffer); } static void print_substream_info(struct snd_info_buffer *buffer, From 8e3bf7cde43339507f9ce15fc10ff329e76ad649 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Wed, 20 Nov 2019 11:49:52 -0600 Subject: [PATCH 121/136] ALSA: aloop: Rename all jiffies timer specific functions This commit does not change the behaviour. It only separates the jiffies timer specific implementation from the generic part. Signed-off-by: Timo Wischer Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-5-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 28 +++++++++++++++------------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 3bfd7c32803c..2f208aaa54cf 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -188,7 +188,7 @@ static inline unsigned int get_rate_shift(struct loopback_pcm *dpcm) } /* call in cable->lock */ -static int loopback_timer_start(struct loopback_pcm *dpcm) +static int loopback_jiffies_timer_start(struct loopback_pcm *dpcm) { unsigned long tick; unsigned int rate_shift = get_rate_shift(dpcm); @@ -209,7 +209,7 @@ static int loopback_timer_start(struct loopback_pcm *dpcm) } /* call in cable->lock */ -static inline int loopback_timer_stop(struct loopback_pcm *dpcm) +static inline int loopback_jiffies_timer_stop(struct loopback_pcm *dpcm) { del_timer(&dpcm->timer); dpcm->timer.expires = 0; @@ -217,7 +217,7 @@ static inline int loopback_timer_stop(struct loopback_pcm *dpcm) return 0; } -static inline int loopback_timer_stop_sync(struct loopback_pcm *dpcm) +static inline int loopback_jiffies_timer_stop_sync(struct loopback_pcm *dpcm) { del_timer_sync(&dpcm->timer); @@ -502,7 +502,8 @@ static inline void bytepos_finish(struct loopback_pcm *dpcm, } /* call in cable->lock */ -static unsigned int loopback_pos_update(struct loopback_cable *cable) +static unsigned int loopback_jiffies_timer_pos_update + (struct loopback_cable *cable) { struct loopback_pcm *dpcm_play = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; @@ -555,14 +556,15 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) return running; } -static void loopback_timer_function(struct timer_list *t) +static void loopback_jiffies_timer_function(struct timer_list *t) { struct loopback_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; spin_lock_irqsave(&dpcm->cable->lock, flags); - if (loopback_pos_update(dpcm->cable) & (1 << dpcm->substream->stream)) { - loopback_timer_start(dpcm); + if (loopback_jiffies_timer_pos_update(dpcm->cable) & + (1 << dpcm->substream->stream)) { + loopback_jiffies_timer_start(dpcm); if (dpcm->period_update_pending) { dpcm->period_update_pending = 0; spin_unlock_irqrestore(&dpcm->cable->lock, flags); @@ -731,18 +733,18 @@ static void free_cable(struct snd_pcm_substream *substream) static int loopback_jiffies_timer_open(struct loopback_pcm *dpcm) { - timer_setup(&dpcm->timer, loopback_timer_function, 0); + timer_setup(&dpcm->timer, loopback_jiffies_timer_function, 0); return 0; } static struct loopback_ops loopback_jiffies_timer_ops = { .open = loopback_jiffies_timer_open, - .start = loopback_timer_start, - .stop = loopback_timer_stop, - .stop_sync = loopback_timer_stop_sync, - .close_substream = loopback_timer_stop_sync, - .pos_update = loopback_pos_update, + .start = loopback_jiffies_timer_start, + .stop = loopback_jiffies_timer_stop, + .stop_sync = loopback_jiffies_timer_stop_sync, + .close_substream = loopback_jiffies_timer_stop_sync, + .pos_update = loopback_jiffies_timer_pos_update, .dpcm_info = loopback_jiffies_timer_dpcm_info, }; From fd1f7c743d30938d3befb1af038a510c003a6a27 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Wed, 20 Nov 2019 11:49:53 -0600 Subject: [PATCH 122/136] ALSA: aloop: Move CABLE_VALID_BOTH to the top of file so all functions can use the same. Signed-off-by: Timo Wischer Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-6-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 2f208aaa54cf..0eacaa9d7862 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -55,6 +55,10 @@ MODULE_PARM_DESC(pcm_notify, "Break capture when PCM format/rate/channels change #define NO_PITCH 100000 +#define CABLE_VALID_PLAYBACK BIT(SNDRV_PCM_STREAM_PLAYBACK) +#define CABLE_VALID_CAPTURE BIT(SNDRV_PCM_STREAM_CAPTURE) +#define CABLE_VALID_BOTH (CABLE_VALID_PLAYBACK | CABLE_VALID_CAPTURE) + struct loopback_cable; struct loopback_pcm; @@ -224,10 +228,6 @@ static inline int loopback_jiffies_timer_stop_sync(struct loopback_pcm *dpcm) return 0; } -#define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) -#define CABLE_VALID_CAPTURE (1 << SNDRV_PCM_STREAM_CAPTURE) -#define CABLE_VALID_BOTH (CABLE_VALID_PLAYBACK|CABLE_VALID_CAPTURE) - static int loopback_check_format(struct loopback_cable *cable, int stream) { struct snd_pcm_runtime *runtime, *cruntime; From 26c53379f98d22d6a3e50bb146651dc7824334d7 Mon Sep 17 00:00:00 2001 From: Timo Wischer Date: Wed, 20 Nov 2019 11:49:54 -0600 Subject: [PATCH 123/136] ALSA: aloop: Support selection of snd_timer instead of jiffies to do synchronous audio forwarding between hardware sound card and aloop devices. Such an audio route could look like the following: Sound card -> Loopback application -> ALSA loop device -> arecord In this case the loopback device should use the sound timer of the sound card. Without this patch the loopback application has to implement an adaptive sample rate converter to align the different clocks of the different ALSA devices. The used timer can be selected by referring to a sound card, its device and subdevice, when loading the module: $ modprobe snd_aloop enable=1 timer_source=[[.[.]]] is the name (id) of the sound card or a card number. and are device and subdevice numbers (defaults are 0). Empty string as a value of timer_source= parameter enables previous functionality (using jiffies timer). Signed-off-by: Timo Wischer Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-7-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 477 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 476 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 0eacaa9d7862..e8a85f887222 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -28,6 +28,7 @@ #include #include #include +#include MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("A loopback soundcard"); @@ -41,6 +42,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; static int pcm_notify[SNDRV_CARDS]; +static char *timer_source[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for loopback soundcard."); @@ -52,6 +54,8 @@ module_param_array(pcm_substreams, int, NULL, 0444); MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-8) for loopback driver."); module_param_array(pcm_notify, int, NULL, 0444); MODULE_PARM_DESC(pcm_notify, "Break capture when PCM format/rate/channels changes."); +module_param_array(timer_source, charp, NULL, 0444); +MODULE_PARM_DESC(timer_source, "Sound card name or number and device/subdevice number of timer to be used. Empty string for jiffies timer [default]."); #define NO_PITCH 100000 @@ -102,6 +106,13 @@ struct loopback_cable { unsigned int pause; /* timer specific */ struct loopback_ops *ops; + /* If sound timer is used */ + struct { + int stream; + struct snd_timer_id id; + struct tasklet_struct event_tasklet; + struct snd_timer_instance *instance; + } snd_timer; }; struct loopback_setup { @@ -122,6 +133,7 @@ struct loopback { struct loopback_cable *cables[MAX_PCM_SUBSTREAMS][2]; struct snd_pcm *pcm[2]; struct loopback_setup setup[MAX_PCM_SUBSTREAMS][2]; + const char *timer_source; }; struct loopback_pcm { @@ -145,6 +157,7 @@ struct loopback_pcm { unsigned int period_size_frac; /* period size in jiffies ticks */ unsigned int last_drift; unsigned long last_jiffies; + /* If jiffies timer is used */ struct timer_list timer; }; @@ -212,6 +225,35 @@ static int loopback_jiffies_timer_start(struct loopback_pcm *dpcm) return 0; } +/* call in cable->lock */ +static int loopback_snd_timer_start(struct loopback_pcm *dpcm) +{ + struct loopback_cable *cable = dpcm->cable; + int err; + + /* Loopback device has to use same period as timer card. Therefore + * wake up for each snd_pcm_period_elapsed() call of timer card. + */ + err = snd_timer_start(cable->snd_timer.instance, 1); + if (err < 0) { + /* do not report error if trying to start but already + * running. For example called by opposite substream + * of the same cable + */ + if (err == -EBUSY) + return 0; + + pcm_err(dpcm->substream->pcm, + "snd_timer_start(%d,%d,%d) failed with %d", + cable->snd_timer.id.card, + cable->snd_timer.id.device, + cable->snd_timer.id.subdevice, + err); + } + + return err; +} + /* call in cable->lock */ static inline int loopback_jiffies_timer_stop(struct loopback_pcm *dpcm) { @@ -221,6 +263,29 @@ static inline int loopback_jiffies_timer_stop(struct loopback_pcm *dpcm) return 0; } +/* call in cable->lock */ +static int loopback_snd_timer_stop(struct loopback_pcm *dpcm) +{ + struct loopback_cable *cable = dpcm->cable; + int err; + + /* only stop if both devices (playback and capture) are not running */ + if (cable->running ^ cable->pause) + return 0; + + err = snd_timer_stop(cable->snd_timer.instance); + if (err < 0) { + pcm_err(dpcm->substream->pcm, + "snd_timer_stop(%d,%d,%d) failed with %d", + cable->snd_timer.id.card, + cable->snd_timer.id.device, + cable->snd_timer.id.subdevice, + err); + } + + return err; +} + static inline int loopback_jiffies_timer_stop_sync(struct loopback_pcm *dpcm) { del_timer_sync(&dpcm->timer); @@ -228,6 +293,30 @@ static inline int loopback_jiffies_timer_stop_sync(struct loopback_pcm *dpcm) return 0; } +/* call in loopback->cable_lock */ +static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm) +{ + struct loopback_cable *cable = dpcm->cable; + + /* snd_timer was not opened */ + if (!cable->snd_timer.instance) + return 0; + + /* wait till drain tasklet has finished if requested */ + tasklet_kill(&cable->snd_timer.event_tasklet); + + /* will only be called from free_cable() when other stream was + * already closed. Other stream cannot be reopened as long as + * loopback->cable_lock is locked. Therefore no need to lock + * cable->lock; + */ + snd_timer_close(cable->snd_timer.instance); + snd_timer_instance_free(cable->snd_timer.instance); + memset(&cable->snd_timer, 0, sizeof(cable->snd_timer)); + + return 0; +} + static int loopback_check_format(struct loopback_cable *cable, int stream) { struct snd_pcm_runtime *runtime, *cruntime; @@ -353,6 +442,13 @@ static void params_change(struct snd_pcm_substream *substream) cable->hw.rate_max = runtime->rate; cable->hw.channels_min = runtime->channels; cable->hw.channels_max = runtime->channels; + + if (cable->snd_timer.instance) { + cable->hw.period_bytes_min = + frames_to_bytes(runtime, runtime->period_size); + cable->hw.period_bytes_max = cable->hw.period_bytes_min; + } + } static int loopback_prepare(struct snd_pcm_substream *substream) @@ -576,6 +672,167 @@ static void loopback_jiffies_timer_function(struct timer_list *t) spin_unlock_irqrestore(&dpcm->cable->lock, flags); } +/* call in cable->lock */ +static int loopback_snd_timer_check_resolution(struct snd_pcm_runtime *runtime, + unsigned long resolution) +{ + if (resolution != runtime->timer_resolution) { + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + /* Worst case estimation of possible values for resolution + * resolution <= (512 * 1024) frames / 8kHz in nsec + * resolution <= 65.536.000.000 nsec + * + * period_size <= 65.536.000.000 nsec / 1000nsec/usec * 192kHz + + * 500.000 + * period_size <= 12.582.912.000.000 <64bit + * / 1.000.000 usec/sec + */ + snd_pcm_uframes_t period_size_usec = + resolution / 1000 * runtime->rate; + /* round to nearest sample rate */ + snd_pcm_uframes_t period_size = + (period_size_usec + 500 * 1000) / (1000 * 1000); + + pcm_err(dpcm->substream->pcm, + "Period size (%lu frames) of loopback device is not corresponding to timer resolution (%lu nsec = %lu frames) of card timer %d,%d,%d. Use period size of %lu frames for loopback device.", + runtime->period_size, resolution, period_size, + cable->snd_timer.id.card, + cable->snd_timer.id.device, + cable->snd_timer.id.subdevice, + period_size); + return -EINVAL; + } + return 0; +} + +static void loopback_snd_timer_period_elapsed(struct loopback_cable *cable, + int event, + unsigned long resolution) +{ + struct loopback_pcm *dpcm_play, *dpcm_capt; + struct snd_pcm_substream *substream_play, *substream_capt; + struct snd_pcm_runtime *valid_runtime; + unsigned int running, elapsed_bytes; + unsigned long flags; + + spin_lock_irqsave(&cable->lock, flags); + running = cable->running ^ cable->pause; + /* no need to do anything if no stream is running */ + if (!running) { + spin_unlock_irqrestore(&cable->lock, flags); + return; + } + + dpcm_play = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; + dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; + substream_play = (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? + dpcm_play->substream : NULL; + substream_capt = (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) ? + dpcm_capt->substream : NULL; + + if (event == SNDRV_TIMER_EVENT_MSTOP) { + if (!dpcm_play || + dpcm_play->substream->runtime->status->state != + SNDRV_PCM_STATE_DRAINING) { + spin_unlock_irqrestore(&cable->lock, flags); + return; + } + } + + valid_runtime = (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? + dpcm_play->substream->runtime : + dpcm_capt->substream->runtime; + + /* resolution is only valid for SNDRV_TIMER_EVENT_TICK events */ + if (event == SNDRV_TIMER_EVENT_TICK) { + /* The hardware rules guarantee that playback and capture period + * are the same. Therefore only one device has to be checked + * here. + */ + if (loopback_snd_timer_check_resolution(valid_runtime, + resolution) < 0) { + spin_unlock_irqrestore(&cable->lock, flags); + if (substream_play) + snd_pcm_stop_xrun(substream_play); + if (substream_capt) + snd_pcm_stop_xrun(substream_capt); + return; + } + } + + elapsed_bytes = frames_to_bytes(valid_runtime, + valid_runtime->period_size); + /* The same timer interrupt is used for playback and capture device */ + if ((running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) && + (running & (1 << SNDRV_PCM_STREAM_CAPTURE))) { + copy_play_buf(dpcm_play, dpcm_capt, elapsed_bytes); + bytepos_finish(dpcm_play, elapsed_bytes); + bytepos_finish(dpcm_capt, elapsed_bytes); + } else if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { + bytepos_finish(dpcm_play, elapsed_bytes); + } else if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { + clear_capture_buf(dpcm_capt, elapsed_bytes); + bytepos_finish(dpcm_capt, elapsed_bytes); + } + spin_unlock_irqrestore(&cable->lock, flags); + + if (substream_play) + snd_pcm_period_elapsed(substream_play); + if (substream_capt) + snd_pcm_period_elapsed(substream_capt); +} + +static void loopback_snd_timer_function(struct snd_timer_instance *timeri, + unsigned long resolution, + unsigned long ticks) +{ + struct loopback_cable *cable = timeri->callback_data; + + loopback_snd_timer_period_elapsed(cable, SNDRV_TIMER_EVENT_TICK, + resolution); +} + +static void loopback_snd_timer_tasklet(unsigned long arg) +{ + struct snd_timer_instance *timeri = (struct snd_timer_instance *)arg; + struct loopback_cable *cable = timeri->callback_data; + + loopback_snd_timer_period_elapsed(cable, SNDRV_TIMER_EVENT_MSTOP, 0); +} + +static void loopback_snd_timer_event(struct snd_timer_instance *timeri, + int event, + struct timespec *tstamp, + unsigned long resolution) +{ + /* Do not lock cable->lock here because timer->lock is already hold. + * There are other functions which first lock cable->lock and than + * timer->lock e.g. + * loopback_trigger() + * spin_lock(&cable->lock) + * loopback_snd_timer_start() + * snd_timer_start() + * spin_lock(&timer->lock) + * Therefore when using the oposit order of locks here it could result + * in a deadlock. + */ + + if (event == SNDRV_TIMER_EVENT_MSTOP) { + struct loopback_cable *cable = timeri->callback_data; + + /* sound card of the timer was stopped. Therefore there will not + * be any further timer callbacks. Due to this forward audio + * data from here if in draining state. When still in running + * state the streaming will be aborted by the usual timeout. It + * should not be aborted here because may be the timer sound + * card does only a recovery and the timer is back soon. + * This tasklet triggers loopback_snd_timer_tasklet() + */ + tasklet_schedule(&cable->snd_timer.event_tasklet); + } +} + static void loopback_jiffies_timer_dpcm_info(struct loopback_pcm *dpcm, struct snd_info_buffer *buffer) { @@ -588,6 +845,20 @@ static void loopback_jiffies_timer_dpcm_info(struct loopback_pcm *dpcm, snd_iprintf(buffer, " timer_expires:\t%lu\n", dpcm->timer.expires); } +static void loopback_snd_timer_dpcm_info(struct loopback_pcm *dpcm, + struct snd_info_buffer *buffer) +{ + struct loopback_cable *cable = dpcm->cable; + + snd_iprintf(buffer, " sound timer:\thw:%d,%d,%d\n", + cable->snd_timer.id.card, + cable->snd_timer.id.device, + cable->snd_timer.id.subdevice); + snd_iprintf(buffer, " timer open:\t\t%s\n", + (cable->snd_timer.stream == SNDRV_PCM_STREAM_CAPTURE) ? + "capture" : "playback"); +} + static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -706,6 +977,23 @@ static int rule_channels(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +static int rule_period_bytes(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; + struct snd_interval t; + + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.period_bytes_min; + t.max = cable->hw.period_bytes_max; + mutex_unlock(&dpcm->loopback->cable_lock); + t.openmin = 0; + t.openmax = 0; + t.integer = 0; + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + static void free_cable(struct snd_pcm_substream *substream) { struct loopback *loopback = substream->private_data; @@ -748,6 +1036,163 @@ static struct loopback_ops loopback_jiffies_timer_ops = { .dpcm_info = loopback_jiffies_timer_dpcm_info, }; +static int loopback_parse_timer_id(const char *str, + struct snd_timer_id *tid) +{ + /* [:](|)[{.,}[{.,}]] */ + const char * const sep_dev = ".,"; + const char * const sep_pref = ":"; + const char *name = str; + char *sep, save = '\0'; + int card_idx = 0, dev = 0, subdev = 0; + int err; + + sep = strpbrk(str, sep_pref); + if (sep) + name = sep + 1; + sep = strpbrk(name, sep_dev); + if (sep) { + save = *sep; + *sep = '\0'; + } + err = kstrtoint(name, 0, &card_idx); + if (err == -EINVAL) { + /* Must be the name, not number */ + for (card_idx = 0; card_idx < snd_ecards_limit; card_idx++) { + struct snd_card *card = snd_card_ref(card_idx); + + if (card) { + if (!strcmp(card->id, name)) + err = 0; + snd_card_unref(card); + } + if (!err) + break; + } + } + if (sep) { + *sep = save; + if (!err) { + char *sep2, save2 = '\0'; + + sep2 = strpbrk(sep + 1, sep_dev); + if (sep2) { + save2 = *sep2; + *sep2 = '\0'; + } + err = kstrtoint(sep + 1, 0, &dev); + if (sep2) { + *sep2 = save2; + if (!err) + err = kstrtoint(sep2 + 1, 0, &subdev); + } + } + } + if (!err && tid) { + tid->card = card_idx; + tid->device = dev; + tid->subdevice = subdev; + } + return err; +} + +/* call in loopback->cable_lock */ +static int loopback_snd_timer_open(struct loopback_pcm *dpcm) +{ + int err = 0; + struct snd_timer_id tid = { + .dev_class = SNDRV_TIMER_CLASS_PCM, + .dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION, + }; + struct snd_timer_instance *timeri; + struct loopback_cable *cable = dpcm->cable; + + spin_lock_irq(&cable->lock); + + /* check if timer was already opened. It is only opened once + * per playback and capture subdevice (aka cable). + */ + if (cable->snd_timer.instance) + goto unlock; + + err = loopback_parse_timer_id(dpcm->loopback->timer_source, &tid); + if (err < 0) { + pcm_err(dpcm->substream->pcm, + "Parsing timer source \'%s\' failed with %d", + dpcm->loopback->timer_source, err); + goto unlock; + } + + cable->snd_timer.stream = dpcm->substream->stream; + cable->snd_timer.id = tid; + + timeri = snd_timer_instance_new(dpcm->loopback->card->id); + if (!timeri) { + err = -ENOMEM; + goto unlock; + } + /* The callback has to be called from another tasklet. If + * SNDRV_TIMER_IFLG_FAST is specified it will be called from the + * snd_pcm_period_elapsed() call of the selected sound card. + * snd_pcm_period_elapsed() helds snd_pcm_stream_lock_irqsave(). + * Due to our callback loopback_snd_timer_function() also calls + * snd_pcm_period_elapsed() which calls snd_pcm_stream_lock_irqsave(). + * This would end up in a dead lock. + */ + timeri->flags |= SNDRV_TIMER_IFLG_AUTO; + timeri->callback = loopback_snd_timer_function; + timeri->callback_data = (void *)cable; + timeri->ccallback = loopback_snd_timer_event; + + /* initialise a tasklet used for draining */ + tasklet_init(&cable->snd_timer.event_tasklet, + loopback_snd_timer_tasklet, (unsigned long)timeri); + + /* snd_timer_close() and snd_timer_open() should not be called with + * locked spinlock because both functions can block on a mutex. The + * mutex loopback->cable_lock is kept locked. Therefore snd_timer_open() + * cannot be called a second time by the other device of the same cable. + * Therefore the following issue cannot happen: + * [proc1] Call loopback_timer_open() -> + * Unlock cable->lock for snd_timer_close/open() call + * [proc2] Call loopback_timer_open() -> snd_timer_open(), + * snd_timer_start() + * [proc1] Call snd_timer_open() and overwrite running timer + * instance + */ + spin_unlock_irq(&cable->lock); + err = snd_timer_open(timeri, &cable->snd_timer.id, current->pid); + spin_lock_irq(&cable->lock); + if (err < 0) { + pcm_err(dpcm->substream->pcm, + "snd_timer_open (%d,%d,%d) failed with %d", + cable->snd_timer.id.card, + cable->snd_timer.id.device, + cable->snd_timer.id.subdevice, + err); + snd_timer_instance_free(timeri); + goto unlock; + } + + cable->snd_timer.instance = timeri; + +unlock: + spin_unlock_irq(&cable->lock); + + return err; +} + +/* stop_sync() is not required for sound timer because it does not need to be + * restarted in loopback_prepare() on Xrun recovery + */ +static struct loopback_ops loopback_snd_timer_ops = { + .open = loopback_snd_timer_open, + .start = loopback_snd_timer_start, + .stop = loopback_snd_timer_stop, + .close_cable = loopback_snd_timer_close_cable, + .dpcm_info = loopback_snd_timer_dpcm_info, +}; + static int loopback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -775,7 +1220,10 @@ static int loopback_open(struct snd_pcm_substream *substream) } spin_lock_init(&cable->lock); cable->hw = loopback_pcm_hardware; - cable->ops = &loopback_jiffies_timer_ops; + if (loopback->timer_source) + cable->ops = &loopback_snd_timer_ops; + else + cable->ops = &loopback_jiffies_timer_ops; loopback->cables[substream->number][dev] = cable; } dpcm->cable = cable; @@ -811,6 +1259,19 @@ static int loopback_open(struct snd_pcm_substream *substream) if (err < 0) goto unlock; + /* In case of sound timer the period time of both devices of the same + * loop has to be the same. + * This rule only takes effect if a sound timer was chosen + */ + if (cable->snd_timer.instance) { + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + rule_period_bytes, dpcm, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, -1); + if (err < 0) + goto unlock; + } + /* loopback_runtime_free() has not to be called if kfree(dpcm) was * already called here. Otherwise it will end up with a double free. */ @@ -1214,6 +1675,18 @@ static int loopback_proc_new(struct loopback *loopback, int cidx) print_cable_info); } +static void loopback_set_timer_source(struct loopback *loopback, + const char *value) +{ + if (loopback->timer_source) { + devm_kfree(loopback->card->dev, loopback->timer_source); + loopback->timer_source = NULL; + } + if (value && *value) + loopback->timer_source = devm_kstrdup(loopback->card->dev, + value, GFP_KERNEL); +} + static int loopback_probe(struct platform_device *devptr) { struct snd_card *card; @@ -1233,6 +1706,8 @@ static int loopback_probe(struct platform_device *devptr) pcm_substreams[dev] = MAX_PCM_SUBSTREAMS; loopback->card = card; + loopback_set_timer_source(loopback, timer_source[dev]); + mutex_init(&loopback->cable_lock); err = loopback_pcm_new(loopback, 0, pcm_substreams[dev]); From c6ae99605633cade7dc61a2b35b04dda90302dad Mon Sep 17 00:00:00 2001 From: Andrew Gabbasov Date: Wed, 20 Nov 2019 11:49:55 -0600 Subject: [PATCH 124/136] ALSA: aloop: Support runtime change of snd_timer via info interface Show and change sound card timer source with read-write info file in proc filesystem. Initial string can still be set as module parameter. The timer source string value can be changed at any time, but it is latched by PCM substream open callback (the first one for a particular cable). At this point it is actually used, that is the string is parsed, and the timer is looked up and opened. The timer source is set for a loopback card (the same as initial setting by module parameter), but every cable uses the value, current at the moment of open. Setting the value to empty string switches the timer to jiffies. Signed-off-by: Andrew Gabbasov Link: https://lore.kernel.org/r/20191120174955.6410-8-andrew_gabbasov@mentor.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 37 ++++++++++++++++++++++++++++++++++--- 1 file changed, 34 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index e8a85f887222..1408403f727a 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1666,7 +1666,7 @@ static void print_cable_info(struct snd_info_entry *entry, mutex_unlock(&loopback->cable_lock); } -static int loopback_proc_new(struct loopback *loopback, int cidx) +static int loopback_cable_proc_new(struct loopback *loopback, int cidx) { char name[32]; @@ -1687,6 +1687,36 @@ static void loopback_set_timer_source(struct loopback *loopback, value, GFP_KERNEL); } +static void print_timer_source_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct loopback *loopback = entry->private_data; + + mutex_lock(&loopback->cable_lock); + snd_iprintf(buffer, "%s\n", + loopback->timer_source ? loopback->timer_source : ""); + mutex_unlock(&loopback->cable_lock); +} + +static void change_timer_source_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct loopback *loopback = entry->private_data; + char line[64]; + + mutex_lock(&loopback->cable_lock); + if (!snd_info_get_line(buffer, line, sizeof(line))) + loopback_set_timer_source(loopback, strim(line)); + mutex_unlock(&loopback->cable_lock); +} + +static int loopback_timer_source_proc_new(struct loopback *loopback) +{ + return snd_card_rw_proc_new(loopback->card, "timer_source", loopback, + print_timer_source_info, + change_timer_source_info); +} + static int loopback_probe(struct platform_device *devptr) { struct snd_card *card; @@ -1719,8 +1749,9 @@ static int loopback_probe(struct platform_device *devptr) err = loopback_mixer_new(loopback, pcm_notify[dev] ? 1 : 0); if (err < 0) goto __nodev; - loopback_proc_new(loopback, 0); - loopback_proc_new(loopback, 1); + loopback_cable_proc_new(loopback, 0); + loopback_cable_proc_new(loopback, 1); + loopback_timer_source_proc_new(loopback); strcpy(card->driver, "Loopback"); strcpy(card->shortname, "Loopback"); sprintf(card->longname, "Loopback %i", dev + 1); From 0dba808eae2627f40024872d2f02f75f0d2b0790 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:01 +0100 Subject: [PATCH 125/136] ALSA: pcm: Introduce managed buffer allocation mode This patch adds the support for the feature to automatically allocate and free PCM buffers, so called "managed buffer allocation" mode. It's set up via new PCM helpers, snd_pcm_set_managed_buffer() and snd_pcm_set_managed_buffer_all(), both of which correspond to the existing preallocator helpers, snd_pcm_lib_preallocate_pages() and snd_pcm_lib_preallocate_pages_for_all(). When the new helper is used, it not only performs the pre-allocation of buffers, but also it manages to call snd_pcm_lib_malloc_pages() before the PCM hw_params ops and snd_lib_pcm_free() after the PCM hw_free ops inside PCM core, respectively. This allows drivers to drop the explicit calls of the memory allocation / release functions, and it will be a good amount of code reduction in the end of this patch series. When the PCM substream is set to the managed buffer allocation mode, the managed_buffer_alloc flag is set in the substream object. Since some drivers want to know when a buffer is newly allocated or re-allocated at hw_params callback (e.g. want to set up the additional stuff for the given buffer only at allocation time), now PCM core turns on buffer_changed flag when the buffer has changed. The standard conversions to use the new API will be straightforward: - Replace snd_pcm_lib_preallocate*() calls with the corresponding snd_pcm_set_managed_buffer*(); the arguments should be unchanged - Drop superfluous snd_pcm_lib_malloc() and snd_pcm_lib_free() calls; the check of snd_pcm_lib_malloc() returns should be replaced with the check of runtime->buffer_changed flag. - If hw_params or hw_free becomes empty, drop them from PCM ops Link: https://lore.kernel.org/r/20191117085308.23915-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 8 ++++ sound/core/pcm_memory.c | 83 ++++++++++++++++++++++++++++++++++------- sound/core/pcm_native.c | 12 ++++++ 3 files changed, 90 insertions(+), 13 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2c0aa884f5f1..253d15c61ce2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -414,6 +414,7 @@ struct snd_pcm_runtime { size_t dma_bytes; /* size of DMA area */ struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + unsigned int buffer_changed:1; /* buffer allocation changed; set only in managed mode */ /* -- audio timestamp config -- */ struct snd_pcm_audio_tstamp_config audio_tstamp_config; @@ -475,6 +476,7 @@ struct snd_pcm_substream { #endif /* CONFIG_SND_VERBOSE_PROCFS */ /* misc flags */ unsigned int hw_opened: 1; + unsigned int managed_buffer_alloc:1; }; #define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0) @@ -1186,6 +1188,12 @@ void snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +void snd_pcm_set_managed_buffer(struct snd_pcm_substream *substream, int type, + struct device *data, size_t size, size_t max); +void snd_pcm_set_managed_buffer_all(struct snd_pcm *pcm, int type, + struct device *data, + size_t size, size_t max); + int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, size_t size, gfp_t gfp_flags); int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 286f333f8e4c..73b770db2372 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -193,9 +193,15 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) /* * pre-allocate the buffer and create a proc file for the substream */ -static void snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream, - size_t size, size_t max) +static void preallocate_pages(struct snd_pcm_substream *substream, + int type, struct device *data, + size_t size, size_t max, bool managed) { + if (snd_BUG_ON(substream->dma_buffer.dev.type)) + return; + + substream->dma_buffer.dev.type = type; + substream->dma_buffer.dev.dev = data; if (size > 0 && preallocate_dma && substream->number < maximum_substreams) preallocate_pcm_pages(substream, size); @@ -205,8 +211,23 @@ static void snd_pcm_lib_preallocate_pages1(struct snd_pcm_substream *substream, substream->dma_max = max; if (max > 0) preallocate_info_init(substream); + if (managed) + substream->managed_buffer_alloc = 1; } +static void preallocate_pages_for_all(struct snd_pcm *pcm, int type, + void *data, size_t size, size_t max, + bool managed) +{ + struct snd_pcm_substream *substream; + int stream; + + for (stream = 0; stream < 2; stream++) + for (substream = pcm->streams[stream].substream; substream; + substream = substream->next) + preallocate_pages(substream, type, data, size, max, + managed); +} /** * snd_pcm_lib_preallocate_pages - pre-allocation for the given DMA type @@ -222,11 +243,7 @@ void snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, int type, struct device *data, size_t size, size_t max) { - if (snd_BUG_ON(substream->dma_buffer.dev.type)) - return; - substream->dma_buffer.dev.type = type; - substream->dma_buffer.dev.dev = data; - snd_pcm_lib_preallocate_pages1(substream, size, max); + preallocate_pages(substream, type, data, size, max, false); } EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages); @@ -245,15 +262,55 @@ void snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int type, void *data, size_t size, size_t max) { - struct snd_pcm_substream *substream; - int stream; - - for (stream = 0; stream < 2; stream++) - for (substream = pcm->streams[stream].substream; substream; substream = substream->next) - snd_pcm_lib_preallocate_pages(substream, type, data, size, max); + preallocate_pages_for_all(pcm, type, data, size, max, false); } EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); +/** + * snd_pcm_set_managed_buffer - set up buffer management for a substream + * @substream: the pcm substream instance + * @type: DMA type (SNDRV_DMA_TYPE_*) + * @data: DMA type dependent data + * @size: the requested pre-allocation size in bytes + * @max: the max. allowed pre-allocation size + * + * Do pre-allocation for the given DMA buffer type, and set the managed + * buffer allocation mode to the given substream. + * In this mode, PCM core will allocate a buffer automatically before PCM + * hw_params ops call, and release the buffer after PCM hw_free ops call + * as well, so that the driver doesn't need to invoke the allocation and + * the release explicitly in its callback. + * When a buffer is actually allocated before the PCM hw_params call, it + * turns on the runtime buffer_changed flag for drivers changing their h/w + * parameters accordingly. + */ +void snd_pcm_set_managed_buffer(struct snd_pcm_substream *substream, int type, + struct device *data, size_t size, size_t max) +{ + preallocate_pages(substream, type, data, size, max, true); +} +EXPORT_SYMBOL(snd_pcm_set_managed_buffer); + +/** + * snd_pcm_set_managed_buffer_all - set up buffer management for all substreams + * for all substreams + * @pcm: the pcm instance + * @type: DMA type (SNDRV_DMA_TYPE_*) + * @data: DMA type dependent data + * @size: the requested pre-allocation size in bytes + * @max: the max. allowed pre-allocation size + * + * Do pre-allocation to all substreams of the given pcm for the specified DMA + * type and size, and set the managed_buffer_alloc flag to each substream. + */ +void snd_pcm_set_managed_buffer_all(struct snd_pcm *pcm, int type, + struct device *data, + size_t size, size_t max) +{ + preallocate_pages_for_all(pcm, type, data, size, max, true); +} +EXPORT_SYMBOL(snd_pcm_set_managed_buffer_all); + #ifdef CONFIG_SND_DMA_SGBUF /* * snd_pcm_sgbuf_ops_page - get the page struct at the given offset diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0c27009dc3df..f1646735bde6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -662,6 +662,14 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) goto _error; + if (substream->managed_buffer_alloc) { + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (err < 0) + goto _error; + runtime->buffer_changed = err > 0; + } + if (substream->ops->hw_params != NULL) { err = substream->ops->hw_params(substream, params); if (err < 0) @@ -723,6 +731,8 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); if (substream->ops->hw_free != NULL) substream->ops->hw_free(substream); + if (substream->managed_buffer_alloc) + snd_pcm_lib_free_pages(substream); return err; } @@ -769,6 +779,8 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) return -EBADFD; if (substream->ops->hw_free) result = substream->ops->hw_free(substream); + if (substream->managed_buffer_alloc) + snd_pcm_lib_free_pages(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); pm_qos_remove_request(&substream->latency_pm_qos_req); return result; From 72b4bcbf1c96c766a0c037a665ef5bbe62bb2628 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:02 +0100 Subject: [PATCH 126/136] ALSA: docs: Update for managed buffer allocation mode Update the documentation for the newly introduced managed buffer allocation mode accordingly. The old preallocation is no longer recommended. Link: https://lore.kernel.org/r/20191117085308.23915-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- .../kernel-api/writing-an-alsa-driver.rst | 99 ++++++++++++------- 1 file changed, 64 insertions(+), 35 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index dcb7940435d9..1086b35db2af 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1270,21 +1270,23 @@ shows only the skeleton, how to build up the PCM interfaces. /* the hardware-specific codes will be here */ .... return 0; - } /* hw_params callback */ static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); + /* the hardware-specific codes will be here */ + .... + return 0; } /* hw_free callback */ static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) { - return snd_pcm_lib_free_pages(substream); + /* the hardware-specific codes will be here */ + .... + return 0; } /* prepare callback */ @@ -1382,9 +1384,9 @@ shows only the skeleton, how to build up the PCM interfaces. &snd_mychip_capture_ops); /* pre-allocation of buffers */ /* NOTE: this may fail */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - &chip->pci->dev, - 64*1024, 64*1024); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pci->dev, + 64*1024, 64*1024); return 0; } @@ -1465,13 +1467,14 @@ The operators are defined typically like this: All the callbacks are described in the Operators_ subsection. After setting the operators, you probably will want to pre-allocate the -buffer. For the pre-allocation, simply call the following: +buffer and set up the managed allocation mode. +For that, simply call the following: :: - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - &chip->pci->dev, - 64*1024, 64*1024); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pci->dev, + 64*1024, 64*1024); It will allocate a buffer up to 64kB as default. Buffer management details will be described in the later section `Buffer and Memory @@ -1621,8 +1624,7 @@ For the operators (callbacks) of each sound driver, most of these records are supposed to be read-only. Only the PCM middle-layer changes / updates them. The exceptions are the hardware description (hw) DMA buffer information and the private data. Besides, if you use the -standard buffer allocation method via -:c:func:`snd_pcm_lib_malloc_pages()`, you don't need to set the +standard managed buffer allocation mode, you don't need to set the DMA buffer information by yourself. In the sections below, important records are explained. @@ -1776,8 +1778,8 @@ the physical address of the buffer. This field is specified only when the buffer is a linear buffer. ``dma_bytes`` holds the size of buffer in bytes. ``dma_private`` is used for the ALSA DMA allocator. -If you use a standard ALSA function, -:c:func:`snd_pcm_lib_malloc_pages()`, for allocating the buffer, +If you use either the managed buffer allocation mode or the standard +API function :c:func:`snd_pcm_lib_malloc_pages()` for allocating the buffer, these fields are set by the ALSA middle layer, and you should *not* change them by yourself. You can read them but not write them. On the other hand, if you want to allocate the buffer by yourself, you'll @@ -1929,8 +1931,12 @@ Many hardware setups should be done in this callback, including the allocation of buffers. Parameters to be initialized are retrieved by -:c:func:`params_xxx()` macros. To allocate buffer, you can call a -helper function, +:c:func:`params_xxx()` macros. + +When you set up the managed buffer allocation mode for the substream, +a buffer is already allocated before this callback gets +called. Alternatively, you can call a helper function below for +allocating the buffer, too. :: @@ -1964,18 +1970,23 @@ hw_free callback static int snd_xxx_hw_free(struct snd_pcm_substream *substream); This is called to release the resources allocated via -``hw_params``. For example, releasing the buffer via -:c:func:`snd_pcm_lib_malloc_pages()` is done by calling the -following: - -:: - - snd_pcm_lib_free_pages(substream); +``hw_params``. This function is always called before the close callback is called. Also, the callback may be called multiple times, too. Keep track whether the resource was already released. +When you have set up the managed buffer allocation mode for the PCM +substream, the allocated PCM buffer will be automatically released +after this callback gets called. Otherwise you'll have to release the +buffer manually. Typically, when the buffer was allocated from the +pre-allocated pool, you can use the standard API function +:c:func:`snd_pcm_lib_malloc_pages()` like: + +:: + + snd_pcm_lib_free_pages(substream); + prepare callback ~~~~~~~~~~~~~~~~ @@ -3543,6 +3554,25 @@ Once the buffer is pre-allocated, you can use the allocator in the Note that you have to pre-allocate to use this function. +Most of drivers use, though, rather the newly introduced "managed +buffer allocation mode" instead of the manual allocation or release. +This is done by calling :c:func:`snd_pcm_set_managed_buffer_all()` +instead of :c:func:`snd_pcm_lib_preallocate_pages_for_all()`. + +:: + + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + &pci->dev, size, max); + +where passed arguments are identical in both functions. +The difference in the managed mode is that PCM core will call +:c:func:`snd_pcm_lib_malloc_pages()` internally already before calling +the PCM ``hw_params`` callback, and call :c:func:`snd_pcm_lib_free_pages()` +after the PCM ``hw_free`` callback automatically. So the driver +doesn't have to call these functions explicitly in its callback any +longer. This made many driver code having NULL ``hw_params`` and +``hw_free`` entries. + External Hardware Buffers ------------------------- @@ -3697,8 +3727,8 @@ provides an interface for handling SG-buffers. The API is provided in ````. For creating the SG-buffer handler, call -:c:func:`snd_pcm_lib_preallocate_pages()` or -:c:func:`snd_pcm_lib_preallocate_pages_for_all()` with +:c:func:`snd_pcm_set_managed_buffer()` or +:c:func:`snd_pcm_set_managed_buffer_all()` with ``SNDRV_DMA_TYPE_DEV_SG`` in the PCM constructor like other PCI pre-allocator. You need to pass ``&pci->dev``, where pci is the :c:type:`struct pci_dev ` pointer of the chip as @@ -3706,8 +3736,8 @@ well. :: - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, - &pci->dev, size, max); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + &pci->dev, size, max); The ``struct snd_sg_buf`` instance is created as ``substream->dma_private`` in turn. You can cast the pointer like: @@ -3716,8 +3746,7 @@ The ``struct snd_sg_buf`` instance is created as struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private; -Then call :c:func:`snd_pcm_lib_malloc_pages()` in the ``hw_params`` -callback as well as in the case of normal PCI buffer. The SG-buffer +Then in :c:func:`snd_pcm_lib_malloc_pages()` call, the common SG-buffer handler will allocate the non-contiguous kernel pages of the given size and map them onto the virtually contiguous memory. The virtual pointer is addressed in runtime->dma_area. The physical address @@ -3726,8 +3755,8 @@ physically non-contiguous. The physical address table is set up in ``sgbuf->table``. You can get the physical address at a certain offset via :c:func:`snd_pcm_sgbuf_get_addr()`. -To release the data, call :c:func:`snd_pcm_lib_free_pages()` in -the ``hw_free`` callback as usual. +If you need to release the SG-buffer data explicitly, call the +standard API function :c:func:`snd_pcm_lib_free_pages()` as usual. Vmalloc'ed Buffers ------------------ @@ -3740,8 +3769,8 @@ buffer preallocation with ``SNDRV_DMA_TYPE_VMALLOC`` type. :: - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, - NULL, 0, 0); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + NULL, 0, 0); The NULL is passed to the device pointer argument, which indicates that the default pages (GFP_KERNEL and GFP_HIGHMEM) will be @@ -3758,7 +3787,7 @@ argument. :: - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC, + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, snd_dma_continuous_data(GFP_KERNEL | __GFP_DMA32), 0, 0); Proc Interface From fc033cbf6fb75452f03774ca5adccac8cf9bc84f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:03 +0100 Subject: [PATCH 127/136] ALSA: pcm: Allow NULL ioctl ops Currently PCM ioctl ops is a mandatory field but almost all drivers simply pass snd_pcm_lib_ioctl. For simplicity, allow to set NULL in the field and call snd_pcm_lib_ioctl() as default. Link: https://lore.kernel.org/r/20191117085308.23915-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f1646735bde6..704ea75199e4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -178,6 +178,16 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore); +/* Run PCM ioctl ops */ +static int snd_pcm_ops_ioctl(struct snd_pcm_substream *substream, + unsigned cmd, void *arg) +{ + if (substream->ops->ioctl) + return substream->ops->ioctl(substream, cmd, arg); + else + return snd_pcm_lib_ioctl(substream, cmd, arg); +} + int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) { struct snd_pcm *pcm = substream->pcm; @@ -448,8 +458,9 @@ static int fixup_unreferenced_params(struct snd_pcm_substream *substream, m = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); i = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); if (snd_mask_single(m) && snd_interval_single(i)) { - err = substream->ops->ioctl(substream, - SNDRV_PCM_IOCTL1_FIFO_SIZE, params); + err = snd_pcm_ops_ioctl(substream, + SNDRV_PCM_IOCTL1_FIFO_SIZE, + params); if (err < 0) return err; } @@ -971,7 +982,7 @@ static int snd_pcm_channel_info(struct snd_pcm_substream *substream, return -EINVAL; memset(info, 0, sizeof(*info)); info->channel = channel; - return substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_CHANNEL_INFO, info); + return snd_pcm_ops_ioctl(substream, SNDRV_PCM_IOCTL1_CHANNEL_INFO, info); } static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, @@ -1647,7 +1658,7 @@ static int snd_pcm_pre_reset(struct snd_pcm_substream *substream, int state) static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; - int err = substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); + int err = snd_pcm_ops_ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); if (err < 0) return err; runtime->hw_ptr_base = 0; From f6161f379c5d455bcedc2cbb1968235a85e151c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:04 +0100 Subject: [PATCH 128/136] ALSA: docs: Update document about the default PCM ioctl ops Mention that it's completely optional now. Link: https://lore.kernel.org/r/20191117085308.23915-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 1086b35db2af..145bf6aad7cb 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -1341,7 +1341,6 @@ shows only the skeleton, how to build up the PCM interfaces. static struct snd_pcm_ops snd_mychip_playback_ops = { .open = snd_mychip_playback_open, .close = snd_mychip_playback_close, - .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mychip_pcm_hw_params, .hw_free = snd_mychip_pcm_hw_free, .prepare = snd_mychip_pcm_prepare, @@ -1353,7 +1352,6 @@ shows only the skeleton, how to build up the PCM interfaces. static struct snd_pcm_ops snd_mychip_capture_ops = { .open = snd_mychip_capture_open, .close = snd_mychip_capture_close, - .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mychip_pcm_hw_params, .hw_free = snd_mychip_pcm_hw_free, .prepare = snd_mychip_pcm_prepare, @@ -1456,7 +1454,6 @@ The operators are defined typically like this: static struct snd_pcm_ops snd_mychip_playback_ops = { .open = snd_mychip_pcm_open, .close = snd_mychip_pcm_close, - .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mychip_pcm_hw_params, .hw_free = snd_mychip_pcm_hw_free, .prepare = snd_mychip_pcm_prepare, @@ -1913,7 +1910,10 @@ ioctl callback ~~~~~~~~~~~~~~ This is used for any special call to pcm ioctls. But usually you can -pass a generic ioctl callback, :c:func:`snd_pcm_lib_ioctl()`. +leave it as NULL, then PCM core calls the generic ioctl callback +function :c:func:`snd_pcm_lib_ioctl()`. If you need to deal with the +unique setup of channel info or reset procedure, you can pass your own +callback function here. hw_params callback ~~~~~~~~~~~~~~~~~~~ From 0821fd77a1129cf4848d82d9275fb4e90e02edf8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 11:05:12 +0100 Subject: [PATCH 129/136] ALSA: pcm: Move PCM_RUNTIME_CHECK() macro into local header It should be used only in the PCM core code locally. Link: https://lore.kernel.org/r/20191117085308.23915-6-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 -- sound/core/pcm_local.h | 2 ++ sound/core/pcm_memory.c | 1 + 3 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 253d15c61ce2..25563317782c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1336,8 +1336,6 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) (IEC958_AES1_CON_PCM_CODER<<8)|\ (IEC958_AES3_CON_FS_48000<<24)) -#define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime) - const char *snd_pcm_format_name(snd_pcm_format_t format); /** diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index 5565e1c4196a..384efd002984 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -72,4 +72,6 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned long offset); #endif +#define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime) + #endif /* __SOUND_CORE_PCM_LOCAL_H */ diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 73b770db2372..d4702cc1d376 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -15,6 +15,7 @@ #include #include #include +#include "pcm_local.h" static int preallocate_dma = 1; module_param(preallocate_dma, int, 0444); From 1e850beea2781d30418743dd99250291cef37919 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:06 +0100 Subject: [PATCH 130/136] ALSA: pcm: Add the support for sync-stop operation The standard programming model of a PCM sound driver is to process snd_pcm_period_elapsed() from an interrupt handler. When a running stream is stopped, PCM core calls the trigger-STOP PCM ops, sets the stream state to SETUP, and moves on to the next step. This is performed in an atomic manner -- this could be called from the interrupt context, after all. The problem is that, if the stream goes further and reaches to the CLOSE state immediately, the stream might be still being processed in snd_pcm_period_elapsed() in the interrupt context, and hits a NULL dereference. Such a crash happens because of the atomic operation, and we can't wait until the stream-stop finishes. For addressing such a problem, this commit adds a new PCM ops, sync_stop. This gets called at the appropriate places that need a sync with the stream-stop, i.e. at hw_params, prepare and hw_free. Some drivers already have a similar mechanism implemented locally, and we'll refactor the code later. Link: https://lore.kernel.org/r/20191117085308.23915-7-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 ++ sound/core/pcm_native.c | 15 +++++++++++++++ 2 files changed, 17 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 25563317782c..8a89fa6fdd5e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -59,6 +59,7 @@ struct snd_pcm_ops { int (*hw_free)(struct snd_pcm_substream *substream); int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); + int (*sync_stop)(struct snd_pcm_substream *substream); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); int (*get_time_info)(struct snd_pcm_substream *substream, struct timespec *system_ts, struct timespec *audio_ts, @@ -395,6 +396,7 @@ struct snd_pcm_runtime { wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; + bool stop_operating; /* sync_stop will be called */ /* -- private section -- */ void *private_data; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 704ea75199e4..163d621ff238 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -568,6 +568,15 @@ static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, #endif } +static void snd_pcm_sync_stop(struct snd_pcm_substream *substream) +{ + if (substream->runtime->stop_operating) { + substream->runtime->stop_operating = false; + if (substream->ops->sync_stop) + substream->ops->sync_stop(substream); + } +} + /** * snd_pcm_hw_param_choose - choose a configuration defined by @params * @pcm: PCM instance @@ -660,6 +669,8 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, if (atomic_read(&substream->mmap_count)) return -EBADFD; + snd_pcm_sync_stop(substream); + params->rmask = ~0U; err = snd_pcm_hw_refine(substream, params); if (err < 0) @@ -788,6 +799,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) snd_pcm_stream_unlock_irq(substream); if (atomic_read(&substream->mmap_count)) return -EBADFD; + snd_pcm_sync_stop(substream); if (substream->ops->hw_free) result = substream->ops->hw_free(substream); if (substream->managed_buffer_alloc) @@ -1313,6 +1325,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) runtime->status->state = state; snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } + runtime->stop_operating = true; wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1589,6 +1602,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) snd_pcm_trigger_tstamp(substream); runtime->status->state = runtime->status->suspended_state; snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); + snd_pcm_sync_stop(substream); } static const struct action_ops snd_pcm_action_resume = { @@ -1709,6 +1723,7 @@ static int snd_pcm_pre_prepare(struct snd_pcm_substream *substream, static int snd_pcm_do_prepare(struct snd_pcm_substream *substream, int state) { int err; + snd_pcm_sync_stop(substream); err = substream->ops->prepare(substream); if (err < 0) return err; From fabb26dcd104027b971c018275fe40f2ebe09ae3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:07 +0100 Subject: [PATCH 131/136] ALSA: pcm: Add card sync_irq field Many PCI and other drivers performs snd_pcm_period_elapsed() simply in its interrupt handler, so the sync_stop operation is just to call synchronize_irq(). Instead of putting this call multiple times, introduce the common card->sync_irq field. When this field is set, PCM core performs synchronize_irq() for sync-stop operation. Each driver just needs to copy its local IRQ number to card->sync_irq, and that's all we need. Link: https://lore.kernel.org/r/20191117085308.23915-8-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/core.h | 1 + sound/core/init.c | 1 + sound/core/pcm_native.c | 2 ++ 3 files changed, 4 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index ee238f100f73..af3dce956c17 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -117,6 +117,7 @@ struct snd_card { struct device card_dev; /* cardX object for sysfs */ const struct attribute_group *dev_groups[4]; /* assigned sysfs attr */ bool registered; /* card_dev is registered? */ + int sync_irq; /* assigned irq, used for PCM sync */ wait_queue_head_t remove_sleep; #ifdef CONFIG_PM diff --git a/sound/core/init.c b/sound/core/init.c index db99b7fad6ad..faa9f03c01ca 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -215,6 +215,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid, init_waitqueue_head(&card->power_sleep); #endif init_waitqueue_head(&card->remove_sleep); + card->sync_irq = -1; device_initialize(&card->card_dev); card->card_dev.parent = parent; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 163d621ff238..1fe581167b7b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -574,6 +574,8 @@ static void snd_pcm_sync_stop(struct snd_pcm_substream *substream) substream->runtime->stop_operating = false; if (substream->ops->sync_stop) substream->ops->sync_stop(substream); + else if (substream->pcm->card->sync_irq > 0) + synchronize_irq(substream->pcm->card->sync_irq); } } From 94722e74272cf4aa08841c98e3a219c7a4c9028b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Nov 2019 09:53:08 +0100 Subject: [PATCH 132/136] ALSA: docs: Update about the new PCM sync_stop ops Add the documentation about the new PCM sync_stop ops and card->sync_irq field. Link: https://lore.kernel.org/r/20191117085308.23915-9-tiwai@suse.de Signed-off-by: Takashi Iwai --- .../kernel-api/writing-an-alsa-driver.rst | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 145bf6aad7cb..f169d58ca019 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -805,6 +805,7 @@ destructor and PCI entries. Example code is shown first, below. return -EBUSY; } chip->irq = pci->irq; + card->sync_irq = chip->irq; /* (2) initialization of the chip hardware */ .... /* (not implemented in this document) */ @@ -965,6 +966,15 @@ usually like the following: return IRQ_HANDLED; } +After requesting the IRQ, you can passed it to ``card->sync_irq`` +field: +:: + + card->irq = chip->irq; + +This allows PCM core automatically performing +:c:func:`synchronize_irq()` at the necessary timing like ``hw_free``. +See the later section `sync_stop callback`_ for details. Now let's write the corresponding destructor for the resources above. The role of destructor is simple: disable the hardware (if already @@ -2059,6 +2069,37 @@ flag set, and you cannot call functions which may sleep. The triggering the DMA. The other stuff should be initialized ``hw_params`` and ``prepare`` callbacks properly beforehand. +sync_stop callback +~~~~~~~~~~~~~~~~~~ + +:: + + static int snd_xxx_sync_stop(struct snd_pcm_substream *substream); + +This callback is optional, and NULL can be passed. It's called after +the PCM core stops the stream and changes the stream state +``prepare``, ``hw_params`` or ``hw_free``. +Since the IRQ handler might be still pending, we need to wait until +the pending task finishes before moving to the next step; otherwise it +might lead to a crash due to resource conflicts or access to the freed +resources. A typical behavior is to call a synchronization function +like :c:func:`synchronize_irq()` here. + +For majority of drivers that need only a call of +:c:func:`synchronize_irq()`, there is a simpler setup, too. +While keeping NULL to ``sync_stop`` PCM callback, the driver can set +``card->sync_irq`` field to store the valid interrupt number after +requesting an IRQ, instead. Then PCM core will look call +:c:func:`synchronize_irq()` with the given IRQ appropriately. + +If the IRQ handler is released at the card destructor, you don't need +to clear ``card->sync_irq``, as the card itself is being released. +So, usually you'll need to add just a single line for assigning +``card->sync_irq`` in the driver code unless the driver re-acquires +the IRQ. When the driver frees and re-acquires the IRQ dynamically +(e.g. for suspend/resume), it needs to clear and re-set +``card->sync_irq`` again appropriately. + pointer callback ~~~~~~~~~~~~~~~~ From aed8c7f40882015aad45088256231babcbc24482 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 21 Nov 2019 10:26:43 +0800 Subject: [PATCH 133/136] ALSA: hda/realtek - Move some alc256 pintbls to fallback table We have a new Dell machine which needs to apply the quirk ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table to fix it this time. And we could remove all pintbls of alc256 for applying DELL1_MIC_NO_PRESENCE on Dell machines. Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20191121022644.8078-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 +++-------------------------------- 1 file changed, 3 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80f66ba85f87..4c83ed4b0d5c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7608,38 +7608,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x1b, 0x01011020}, {0x21, 0x02211010}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60130}, - {0x14, 0x90170110}, - {0x1b, 0x01011020}, - {0x21, 0x0221101f}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60160}, - {0x14, 0x90170120}, - {0x21, 0x02211030}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60170}, - {0x14, 0x90170120}, - {0x21, 0x02211030}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell Inspiron 5468", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60180}, - {0x14, 0x90170120}, - {0x21, 0x02211030}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0xb7a60130}, - {0x14, 0x90170110}, - {0x21, 0x02211020}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60130}, - {0x14, 0x90170110}, - {0x14, 0x01011020}, - {0x21, 0x0221101f}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x14, 0x90170110}, - {0x1b, 0x01011020}, - {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC, {0x14, 0x90170110}, {0x1b, 0x90a70130}, @@ -7852,6 +7820,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1b, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x19, 0x40000000}, + {0x1a, 0x40000000}), {} }; From d64ebdbfd4f71406f58210f5ccb16977b4cd31d2 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 21 Nov 2019 10:26:44 +0800 Subject: [PATCH 134/136] ALSA: hda/realtek - Move some alc236 pintbls to fallback table We have a new Dell machine which needs to apply the quirk ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table to fix it this time. And we could remove all pintbls of alc236 for applying DELL1_MIC_NO_PRESENCE on Dell machines. Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20191121022644.8078-2-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 +++-------------- 1 file changed, 3 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c83ed4b0d5c..4ebe104cb592 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7512,20 +7512,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x19, 0x02a11020}, {0x1a, 0x02a11030}, {0x21, 0x0221101f}), - SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60140}, - {0x14, 0x90170110}, - {0x21, 0x02211020}), - SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60140}, - {0x14, 0x90170150}, - {0x21, 0x02211020}), - SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x21, 0x02211020}), - SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, - {0x14, 0x90170110}, - {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, {0x21, 0x02211020}), @@ -7823,6 +7809,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x19, 0x40000000}, + {0x1a, 0x40000000}), {} }; From 695d1ec3994f9de2cefae80ee2087c95d2e5a2f3 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 21 Nov 2019 10:54:27 +0800 Subject: [PATCH 135/136] ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop The headset on this machine is not defined, after applying the quirk ALC256_FIXUP_ASUS_HEADSET_MIC, the headset-mic works well BugLink: https://bugs.launchpad.net/bugs/1846148 Cc: Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20191121025427.8856-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4ebe104cb592..bd0c767981b1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7248,6 +7248,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), #if 0 From 6b5a6bccec745ea813cab40022b9fc970b535680 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 11 Nov 2019 00:13:56 +1030 Subject: [PATCH 136/136] ALSA: usb-audio: Fix Scarlett 6i6 Gen 2 port data The s6i6_gen2_info.ports[] array had the Mixer and PCM port type entries in the wrong place. Use designators to explicitly specify the array elements being set. Signed-off-by: Geoffrey D. Bennett Tested-by: Alex Fellows Tested-by: Markus Schroetter Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface") Link: https://lore.kernel.org/r/20191110134356.GA31589@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett_gen2.c | 36 ++++++++++++++++----------------- 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 7d460b1f1735..94b903d95afa 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -261,34 +261,34 @@ static const struct scarlett2_device_info s6i6_gen2_info = { }, .ports = { - { + [SCARLETT2_PORT_TYPE_NONE] = { .id = 0x000, .num = { 1, 0, 8, 8, 8 }, .src_descr = "Off", .src_num_offset = 0, }, - { + [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, .num = { 4, 4, 4, 4, 4 }, .src_descr = "Analogue %d", .src_num_offset = 1, .dst_descr = "Analogue Output %02d Playback" }, - { + [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, .dst_descr = "S/PDIF Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_MIX] = { .id = 0x300, .num = { 10, 18, 18, 18, 18 }, .src_descr = "Mix %c", .src_num_offset = 65, .dst_descr = "Mixer Input %02d Capture" }, - { + [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, .num = { 6, 6, 6, 6, 6 }, .src_descr = "PCM %d", @@ -317,44 +317,44 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, .ports = { - { + [SCARLETT2_PORT_TYPE_NONE] = { .id = 0x000, .num = { 1, 0, 8, 8, 4 }, .src_descr = "Off", .src_num_offset = 0, }, - { + [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, .num = { 8, 6, 6, 6, 6 }, .src_descr = "Analogue %d", .src_num_offset = 1, .dst_descr = "Analogue Output %02d Playback" }, - { + [SCARLETT2_PORT_TYPE_SPDIF] = { + .id = 0x180, /* S/PDIF outputs aren't available at 192KHz * but are included in the USB mux I/O * assignment message anyway */ - .id = 0x180, .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, .dst_descr = "S/PDIF Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_ADAT] = { .id = 0x200, .num = { 8, 0, 0, 0, 0 }, .src_descr = "ADAT %d", .src_num_offset = 1, }, - { + [SCARLETT2_PORT_TYPE_MIX] = { .id = 0x300, .num = { 10, 18, 18, 18, 18 }, .src_descr = "Mix %c", .src_num_offset = 65, .dst_descr = "Mixer Input %02d Capture" }, - { + [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, .num = { 20, 18, 18, 14, 10 }, .src_descr = "PCM %d", @@ -387,20 +387,20 @@ static const struct scarlett2_device_info s18i20_gen2_info = { }, .ports = { - { + [SCARLETT2_PORT_TYPE_NONE] = { .id = 0x000, .num = { 1, 0, 8, 8, 6 }, .src_descr = "Off", .src_num_offset = 0, }, - { + [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, .num = { 8, 10, 10, 10, 10 }, .src_descr = "Analogue %d", .src_num_offset = 1, .dst_descr = "Analogue Output %02d Playback" }, - { + [SCARLETT2_PORT_TYPE_SPDIF] = { /* S/PDIF outputs aren't available at 192KHz * but are included in the USB mux I/O * assignment message anyway @@ -411,21 +411,21 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .src_num_offset = 1, .dst_descr = "S/PDIF Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_ADAT] = { .id = 0x200, .num = { 8, 8, 8, 4, 0 }, .src_descr = "ADAT %d", .src_num_offset = 1, .dst_descr = "ADAT Output %d Playback" }, - { + [SCARLETT2_PORT_TYPE_MIX] = { .id = 0x300, .num = { 10, 18, 18, 18, 18 }, .src_descr = "Mix %c", .src_num_offset = 65, .dst_descr = "Mixer Input %02d Capture" }, - { + [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, .num = { 20, 18, 18, 14, 10 }, .src_descr = "PCM %d",