From 6ff674615c91d942e4cfb4880c14733b616c9032 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 30 Mar 2011 08:24:00 +0200 Subject: [PATCH 1/8] ALSA: firewire-speakers: fix hang when unplugging a running device When aborting a PCM stream, the xrun is signaled only if the stream is running. When disconnecting a PCM stream, calling snd_card_disconnect() too early would change the stream into a non-running state and thus prevent the xrun from being noticed by user space. To prevent this, move the snd_card_disconnect() call after the xrun. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/speakers.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index 0fce9218abb1..5466de8527bd 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -778,10 +778,9 @@ static int __devexit fwspk_remove(struct device *dev) { struct fwspk *fwspk = dev_get_drvdata(dev); - snd_card_disconnect(fwspk->card); - mutex_lock(&fwspk->mutex); amdtp_out_stream_pcm_abort(&fwspk->stream); + snd_card_disconnect(fwspk->card); fwspk_stop_stream(fwspk); mutex_unlock(&fwspk->mutex); From 6ebb8a4a43e34f999ab36f27f972f3cd751cda4f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 30 Mar 2011 08:24:25 +0200 Subject: [PATCH 2/8] ALSA: ens1371: fix Creative Ectiva support To make the EV1938 chip work, add a magic bit and an extra delay. Signed-off-by: Clemens Ladisch Tested-by: Tino Schmidt Cc: all 2.6.x Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 537cfba829a5..863eafea691f 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -229,6 +229,7 @@ MODULE_PARM_DESC(lineio, "Line In to Rear Out (0 = auto, 1 = force)."); #define ES_REG_1371_CODEC 0x14 /* W/R: Codec Read/Write register address */ #define ES_1371_CODEC_RDY (1<<31) /* codec ready */ #define ES_1371_CODEC_WIP (1<<30) /* codec register access in progress */ +#define EV_1938_CODEC_MAGIC (1<<26) #define ES_1371_CODEC_PIRD (1<<23) /* codec read/write select register */ #define ES_1371_CODEC_WRITE(a,d) ((((a)&0x7f)<<16)|(((d)&0xffff)<<0)) #define ES_1371_CODEC_READS(a) ((((a)&0x7f)<<16)|ES_1371_CODEC_PIRD) @@ -603,12 +604,18 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531, #ifdef CHIP1371 +static inline bool is_ev1938(struct ensoniq *ensoniq) +{ + return ensoniq->pci->device == 0x8938; +} + static void snd_es1371_codec_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct ensoniq *ensoniq = ac97->private_data; - unsigned int t, x; + unsigned int t, x, flag; + flag = is_ev1938(ensoniq) ? EV_1938_CODEC_MAGIC : 0; mutex_lock(&ensoniq->src_mutex); for (t = 0; t < POLL_COUNT; t++) { if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) { @@ -630,7 +637,8 @@ static void snd_es1371_codec_write(struct snd_ac97 *ac97, 0x00010000) break; } - outl(ES_1371_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1371_CODEC)); + outl(ES_1371_CODEC_WRITE(reg, val) | flag, + ES_REG(ensoniq, 1371_CODEC)); /* restore SRC reg */ snd_es1371_wait_src_ready(ensoniq); outl(x, ES_REG(ensoniq, 1371_SMPRATE)); @@ -647,8 +655,9 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, unsigned short reg) { struct ensoniq *ensoniq = ac97->private_data; - unsigned int t, x, fail = 0; + unsigned int t, x, flag, fail = 0; + flag = is_ev1938(ensoniq) ? EV_1938_CODEC_MAGIC : 0; __again: mutex_lock(&ensoniq->src_mutex); for (t = 0; t < POLL_COUNT; t++) { @@ -671,7 +680,8 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, 0x00010000) break; } - outl(ES_1371_CODEC_READS(reg), ES_REG(ensoniq, 1371_CODEC)); + outl(ES_1371_CODEC_READS(reg) | flag, + ES_REG(ensoniq, 1371_CODEC)); /* restore SRC reg */ snd_es1371_wait_src_ready(ensoniq); outl(x, ES_REG(ensoniq, 1371_SMPRATE)); @@ -683,6 +693,11 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, /* now wait for the stinkin' data (RDY) */ for (t = 0; t < POLL_COUNT; t++) { if ((x = inl(ES_REG(ensoniq, 1371_CODEC))) & ES_1371_CODEC_RDY) { + if (is_ev1938(ensoniq)) { + for (t = 0; t < 100; t++) + inl(ES_REG(ensoniq, CONTROL)); + x = inl(ES_REG(ensoniq, 1371_CODEC)); + } mutex_unlock(&ensoniq->src_mutex); return ES_1371_CODEC_READ(x); } From 840126579da56edae8ecc4a0d85198f742982f10 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 31 Mar 2011 09:36:19 +0200 Subject: [PATCH 3/8] ALSA: HDA: Add dock mic quirk for Lenovo Thinkpad X220 This quirk is needed for the docking station mic of Lenovo Thinkpad X220 to function correctly. BugLink: http://bugs.launchpad.net/bugs/746259 Cc: stable@kernel.org Tested-by: James Ferguson Signed-off-by: David Henningsson 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 d08cf31596f3..69e33869a53e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3034,6 +3034,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} From 12ff414e2e4512f59fe191dc18e856e2939a1c79 Mon Sep 17 00:00:00 2001 From: Kelly Anderson Date: Fri, 1 Apr 2011 11:58:25 +0200 Subject: [PATCH 4/8] ALSA: pcm: fix infinite loop in snd_pcm_update_hw_ptr0() When period interrupts are disabled, snd_pcm_update_hw_ptr0() compares the current time against the time estimated for the current hardware pointer to detect xruns. The somewhat fuzzy threshold in the while loop makes it possible that hdelta becomes negative; the comparison being done with unsigned types then makes the loop go through the entire 263 negative range, and, depending on the value, never reach an unsigned value that is small enough to stop the loop. Doing this with interrupts disabled results in the machine locking up. To prevent this, ensure that the loop condition uses signed types for both operands so that the comparison is correctly done. Many thanks to Kelly Anderson for debugging this. Reported-by: Nix Reported-by: "Christopher K." Reported-and-tested-by: Kelly Anderson Signed-off-by: Kelly Anderson [cl: remove unneeded casts; use a temp variable] Signed-off-by: Clemens Ladisch Cc: 2.6.38 Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a82e3756a72d..64449cb8f873 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -375,6 +375,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } if (runtime->no_period_wakeup) { + snd_pcm_sframes_t xrun_threshold; /* * Without regular period interrupts, we have to check * the elapsed time to detect xruns. @@ -383,7 +384,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (jdelta < runtime->hw_ptr_buffer_jiffies / 2) goto no_delta_check; hdelta = jdelta - delta * HZ / runtime->rate; - while (hdelta > runtime->hw_ptr_buffer_jiffies / 2 + 1) { + xrun_threshold = runtime->hw_ptr_buffer_jiffies / 2 + 1; + while (hdelta > xrun_threshold) { delta += runtime->buffer_size; hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) From b2cb1292b1c7c73abbdc0e07ef3aab056fc2615f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 5 Apr 2011 07:55:24 +0200 Subject: [PATCH 5/8] ALSA: HDA: Fix dock mic for Lenovo X220-tablet Without the "thinkpad" quirk, the dock mic in Lenovo X220 tablet edition won't work. BugLink: http://bugs.launchpad.net/bugs/751033 Cc: stable@kernel.org Tested-by: James Ferguson Signed-off-by: David Henningsson 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 69e33869a53e..ad97d937d3a8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3035,6 +3035,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} From 49c039f071d36586ba32da75996ef339e4ab8405 Mon Sep 17 00:00:00 2001 From: Tarek Soliman Date: Mon, 4 Apr 2011 09:23:53 -0500 Subject: [PATCH 6/8] ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable There are many USB MIDI cables out there that have buggy firmware that reports it can do more than 4 bytes in a packet when they can only properly handle 4 This patch adds the ID of yet another one of those cables Signed-off-by: Tarek Soliman Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b4b39c0b6c9e..f9289102886a 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1301,6 +1301,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ case USB_ID(0x15ca, 0x1806): /* Textech USB Midi Cable */ case USB_ID(0x1a86, 0x752d): /* QinHeng CH345 "USB2.0-MIDI" */ + case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */ ep->max_transfer = 4; break; /* From 1f348522844bb1f6e7b10d50b9e8aa89a2511b09 Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Wed, 6 Apr 2011 17:19:04 -0700 Subject: [PATCH 7/8] ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums The MCP7x hardware computes the audio infoframe channel count automatically, but requires the audio driver to set the audio infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum control verb. When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum to (0x71 - chan - chanmask). For example, for 2ch audio, chan == 1 and chanmask == 0 so the checksum is set to 0x70. When audio playback finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the channel formats, causing the channel count to revert to 8ch. Since the checksum is not reset, the hardware starts generating audio infoframes with invalid checksums. This causes some displays to blank the video. Fix this by updating the checksum and channel mask when the device is closed and also when it is first initialized. In addition, make sure that the channel mask is appropriate for an 8ch infoframe by setting it to 0x13 (FL FR LFE FC RL RR RLC RRC). Signed-off-by: Aaron Plattner Acked-by: Stephen Warren Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 70 ++++++++++++++++++++++++-------------- 1 file changed, 44 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 251773e45f61..715615a88a8d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1280,6 +1280,39 @@ static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec, + int channels) +{ + unsigned int chanmask; + int chan = channels ? (channels - 1) : 1; + + switch (channels) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + + /* Set the audio infoframe channel allocation and checksum fields. The + * channel count is computed implicitly by the hardware. */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); +} + static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1298,6 +1331,10 @@ static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, AC_VERB_SET_STREAM_FORMAT, 0); } + /* The audio hardware sends a channel count of 0x7 (8ch) when all the + * streams are disabled. */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -1308,37 +1345,16 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { int chs; - unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + unsigned int dataDCC1, dataDCC2, channel_id; int i; mutex_lock(&codec->spdif_mutex); chs = substream->runtime->channels; - chan = chs ? (chs - 1) : 1; - switch (chs) { - default: - case 0: - case 2: - chanmask = 0x00; - break; - case 4: - chanmask = 0x08; - break; - case 6: - chanmask = 0x0b; - break; - case 8: - chanmask = 0x13; - break; - } dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; dataDCC2 = 0x2; - /* set the Audio InforFrame Channel Allocation */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Channel_Allocation, chanmask); - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) snd_hda_codec_write(codec, @@ -1413,10 +1429,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, } } - /* set the Audio Info Frame Checksum */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Info_Frame_Checksum, - (0x71 - chan - chanmask)); + nvhdmi_8ch_7x_set_info_frame_parameters(codec, chs); mutex_unlock(&codec->spdif_mutex); return 0; @@ -1512,6 +1525,11 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) spec->multiout.max_channels = 8; spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + + /* Initialize the audio infoframe channel mask and checksum to something + * valid */ + nvhdmi_8ch_7x_set_info_frame_parameters(codec, 8); + return 0; } From 262ac22d21ee2bf3e1655b2e5e45cc94b356e62f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 7 Apr 2011 11:43:00 +0200 Subject: [PATCH 8/8] ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E) In cases where there is only one internal mic connected to ADC 0x11, alc275_setup_dual_adc won't handle the case, so we need to add the ADC node to the array of candidates. Cc: stable@kernel.org BugLink: http://bugs.launchpad.net/bugs/752792 Reported-by: Vincenzo Pii Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f4508c54..d9f1ef7dac2e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14124,7 +14124,7 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { }; static hda_nid_t alc269_adc_candidates[] = { - 0x08, 0x09, 0x07, + 0x08, 0x09, 0x07, 0x11, }; #define alc269_modes alc260_modes