From f373a811fd9a69fc8bafb9bcb41d2cfa36c62665 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 11 Dec 2020 13:06:52 +0300 Subject: [PATCH 01/16] ASoC: Intel: fix error code cnl_set_dsp_D0() Return -ETIMEDOUT if the dsp boot times out instead of returning success. Fixes: cb6a55284629 ("ASoC: Intel: cnl: Add sst library functions for cnl platform") Signed-off-by: Dan Carpenter Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/X9NEvCzuN+IObnTN@mwanda Signed-off-by: Mark Brown --- sound/soc/intel/skylake/cnl-sst.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index fcd8dff27ae8..1275c149acc0 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -224,6 +224,7 @@ static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) "dsp boot timeout, status=%#x error=%#x\n", sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS), sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE)); + ret = -ETIMEDOUT; goto err; } } else { From fe6ce6c394fb1ef1d8a6384c5180e70893157f22 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 15 Dec 2020 15:05:11 +0200 Subject: [PATCH 02/16] MAINTAINERS: Update email address for TI ASoC and twl4030 codec drivers My employment with TI is coming to an end, it is my intention to look after the drivers I have worked with over the years. Signed-off-by: Peter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20201215130512.8753-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- MAINTAINERS | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 3da6d8c154e4..666b76a634a8 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -12644,7 +12644,7 @@ F: include/misc/ocxl* F: include/uapi/misc/ocxl.h OMAP AUDIO SUPPORT -M: Peter Ujfalusi +M: Peter Ujfalusi M: Jarkko Nikula L: alsa-devel@alsa-project.org (moderated for non-subscribers) L: linux-omap@vger.kernel.org @@ -17280,7 +17280,7 @@ F: arch/xtensa/ F: drivers/irqchip/irq-xtensa-* TEXAS INSTRUMENTS ASoC DRIVERS -M: Peter Ujfalusi +M: Peter Ujfalusi L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/ti/ @@ -17571,7 +17571,7 @@ F: Documentation/devicetree/bindings/net/nfc/trf7970a.txt F: drivers/nfc/trf7970a.c TI TWL4030 SERIES SOC CODEC DRIVER -M: Peter Ujfalusi +M: Peter Ujfalusi L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/twl4030* From 61fc03b6512b18f27a25002426d595f5a36645ed Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 15 Dec 2020 15:05:12 +0200 Subject: [PATCH 03/16] ASoC: dt-bindings: ti, j721e: Update maintainer and author information My employment with TI is coming to an end, add the copyright and author comments as they due and change the maintainer mail address. Signed-off-by: Peter Ujfalusi Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20201215130512.8753-3-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,j721e-cpb-audio.yaml | 4 +++- .../devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml index 805da4d6a88e..ec06789b21df 100644 --- a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml @@ -1,4 +1,6 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi %YAML 1.2 --- $id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml# @@ -7,7 +9,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Texas Instruments J721e Common Processor Board Audio Support maintainers: - - Peter Ujfalusi + - Peter Ujfalusi description: | The audio support on the board is using pcm3168a codec connected to McASP10 diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml index bb780f621628..ee9f960de36b 100644 --- a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml @@ -1,4 +1,6 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +# Author: Peter Ujfalusi %YAML 1.2 --- $id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml# @@ -7,7 +9,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Texas Instruments J721e Common Processor Board Audio Support maintainers: - - Peter Ujfalusi + - Peter Ujfalusi description: | The Infotainment board plugs into the Common Processor Board, the support of the From 5c6679b5cb120f07652418524ab186ac47680b49 Mon Sep 17 00:00:00 2001 From: Thomas Hebb Date: Sat, 12 Dec 2020 17:20:12 -0800 Subject: [PATCH 04/16] ASoC: dapm: remove widget from dirty list on free A widget's "dirty" list_head, much like its "list" list_head, eventually chains back to a list_head on the snd_soc_card itself. This means that the list can stick around even after the widget (or all widgets) have been freed. Currently, however, widgets that are in the dirty list when freed remain there, corrupting the entire list and leading to memory errors and undefined behavior when the list is next accessed or modified. I encountered this issue when a component failed to probe relatively late in snd_soc_bind_card(), causing it to bail out and call soc_cleanup_card_resources(), which eventually called snd_soc_dapm_free() with widgets that were still dirty from when they'd been added. Fixes: db432b414e20 ("ASoC: Do DAPM power checks only for widgets changed since last run") Cc: stable@vger.kernel.org Signed-off-by: Thomas Hebb Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/f8b5f031d50122bf1a9bfc9cae046badf4a7a31a.1607822410.git.tommyhebb@gmail.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f0c86cbdcca..2b75d0139e47 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2486,6 +2486,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) enum snd_soc_dapm_direction dir; list_del(&w->list); + list_del(&w->dirty); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both From 4ad2d3cf2a299645bdc6d72e5b8ee11b2ed147ac Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 16 Dec 2020 11:28:59 +0000 Subject: [PATCH 05/16] ASoC: codecs: fix spelling mistake in Kconfig "comunicate" -> "communicate" There is a spelling mistake in the Kconfig help text. Fix it. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20201216112859.11564-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5e4e68112791..ac63e7c176c1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -458,7 +458,7 @@ config SND_SOC_ADAU7118_HW help Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter. In this mode, the device works in standalone mode which - means that there is no bus to comunicate with it. Stereo mode is not + means that there is no bus to communicate with it. Stereo mode is not supported in this mode. To compile this driver as a module, choose M here: the module From e49037ad12e47cd34239b99b010c5438844923af Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 16 Dec 2020 12:59:13 +0000 Subject: [PATCH 06/16] ASoC: SOF: Fix spelling mistake in Kconfig "ond" -> "and" There is a spelling mistake in the Kconfig help text. Fix it. Signed-off-by: Colin Ian King Acked-by: Kai Vehmanen Link: https://lore.kernel.org/r/20201216125913.16041-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 031dad5fc4c7..3e8b6c035ce3 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -122,7 +122,7 @@ config SND_SOC_SOF_DEBUG_XRUN_STOP bool "SOF stop on XRUN" help This option forces PCMs to stop on any XRUN event. This is useful to - preserve any trace data ond pipeline status prior to the XRUN. + preserve any trace data and pipeline status prior to the XRUN. Say Y if you are debugging SOF FW pipeline XRUNs. If unsure select "N". From acd894aee3149c15847bc4f0690fccba59ced5e7 Mon Sep 17 00:00:00 2001 From: shengjiu wang Date: Wed, 16 Dec 2020 18:44:24 +0800 Subject: [PATCH 07/16] ASoC: imx-hdmi: Fix warning of the uninitialized variable ret When condition ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) is true, then goto fail, the uninitialized variable ret will be returned. Signed-off-by: shengjiu wang Reported-by: kernel test robot Acked-by: Nicolin Chen Fixes: 6a5f850aa83a ("ASoC: fsl: Add imx-hdmi machine driver") Reviewed-by: Fabio Estevam Link: https://lore.kernel.org/r/1608115464-18710-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 2c2a76a71940..ede4a9ad1054 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev) if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) { dev_err(&pdev->dev, "Invalid HDMI DAI link\n"); + ret = -EINVAL; goto fail; } From 13733775326ea9eb81c6148ad60c43b8d231a343 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 16 Dec 2020 11:26:08 +0000 Subject: [PATCH 08/16] ASoC: atmel: fix spelling mistake in Kconfig "programable" -> "programmable" There are a couple of spelling mistakes in the Kconfig help text. Fix them. Signed-off-by: Colin Ian King Reviewed-by: Codrin Ciubotariu Link: https://lore.kernel.org/r/20201216112608.11385-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 142373ec411a..9fe9471f4514 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -143,7 +143,7 @@ config SND_MCHP_SOC_SPDIFTX - sama7g5 This S/PDIF TX driver is compliant with IEC-60958 standard and - includes programable User Data and Channel Status fields. + includes programmable User Data and Channel Status fields. config SND_MCHP_SOC_SPDIFRX tristate "Microchip ASoC driver for boards using S/PDIF RX" @@ -157,5 +157,5 @@ config SND_MCHP_SOC_SPDIFRX - sama7g5 This S/PDIF RX driver is compliant with IEC-60958 standard and - includes programable User Data and Channel Status fields. + includes programmable User Data and Channel Status fields. endif From 315fbe4cef98ee5fb6085bc54c7f25eb06466c70 Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 17 Dec 2020 13:38:33 +0530 Subject: [PATCH 09/16] ASoC: qcom: Fix incorrect volatile registers MI2S and DMA control registers are not volatile, so remove these from volatile registers list. Registers reset state check by reading non volatile registers makes no use, so remove error check from cpu and platform trigger callbacks. Initialized map variable two times in lpass platform trigger API, so remove redundant initialization. Fixes commit b1824968221cc ("ASoC: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state") Signed-off-by: V Sujith Kumar Reddy Signed-off-by: Srinivasa Rao Mandadapu Link: https://lore.kernel.org/r/1608192514-29695-2-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 20 ++------------------ sound/soc/qcom/lpass-platform.c | 15 --------------- 2 files changed, 2 insertions(+), 33 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd19ab9..c5e99c2d89c7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -270,18 +270,6 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; unsigned int id = dai->driver->id; int ret = -EINVAL; - unsigned int val = 0; - - ret = regmap_read(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), &val); - if (ret) { - dev_err(dai->dev, "error reading from i2sctl reg: %d\n", ret); - return ret; - } - if (val == LPAIF_I2SCTL_RESET_STATE) { - dev_err(dai->dev, "error in i2sctl register state\n"); - return -ENOTRECOVERABLE; - } switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -454,20 +442,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09dede5f9..232deee6fde5 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -452,7 +452,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, unsigned int reg_irqclr = 0, val_irqclr = 0; unsigned int reg_irqen = 0, val_irqen = 0, val_mask = 0; unsigned int dai_id = cpu_dai->driver->id; - unsigned int dma_ctrl_reg = 0; ch = pcm_data->dma_ch; if (dir == SNDRV_PCM_STREAM_PLAYBACK) { @@ -469,17 +468,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, id = pcm_data->dma_ch - v->wrdma_channel_start; map = drvdata->lpaif_map; } - ret = regmap_read(map, LPAIF_DMACTL_REG(v, ch, dir, dai_id), &dma_ctrl_reg); - if (ret) { - dev_err(soc_runtime->dev, "error reading from rdmactl reg: %d\n", ret); - return ret; - } - if (dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE || - dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE + 1) { - dev_err(soc_runtime->dev, "error in rdmactl register state\n"); - return -ENOTRECOVERABLE; - } switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -500,7 +489,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, "error writing to rdmactl reg: %d\n", ret); return ret; } - map = drvdata->hdmiif_map; reg_irqclr = LPASS_HDMITX_APP_IRQCLEAR_REG(v); val_irqclr = (LPAIF_IRQ_ALL(ch) | LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) | @@ -519,7 +507,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; case MI2S_PRIMARY: case MI2S_SECONDARY: - map = drvdata->lpaif_map; reg_irqclr = LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST); val_irqclr = LPAIF_IRQ_ALL(ch); @@ -563,7 +550,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, "error writing to rdmactl reg: %d\n", ret); return ret; } - map = drvdata->hdmiif_map; reg_irqen = LPASS_HDMITX_APP_IRQEN_REG(v); val_mask = (LPAIF_IRQ_ALL(ch) | LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) | @@ -573,7 +559,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; case MI2S_PRIMARY: case MI2S_SECONDARY: - map = drvdata->lpaif_map; reg_irqen = LPAIF_IRQEN_REG(v, LPAIF_IRQ_PORT_HOST); val_mask = LPAIF_IRQ_ALL(ch); val_irqen = 0; From 8d1bfc04c97407767559f6389a0f0fb060cbe25e Mon Sep 17 00:00:00 2001 From: Srinivasa Rao Mandadapu Date: Thu, 17 Dec 2020 13:38:34 +0530 Subject: [PATCH 10/16] ASoC: qcom: Add support for playback recover after resume To support playback continuation after hard suspend(bypass powerd) and resume do regcache sync with component driver pm ops. Signed-off-by: V Sujith Kumar Reddy Signed-off-by: Srinivasa Rao Mandadapu Reviewed-by: Srinivas Kandagatla Tested-by: Steev Klimaszewski Link: https://lore.kernel.org/r/1608192514-29695-3-git-send-email-srivasam@codeaurora.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 35 +++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 232deee6fde5..d1c248590f3a 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -823,6 +823,39 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component, } } +static int lpass_platform_pcmops_suspend(struct snd_soc_component *component) +{ + struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); + struct regmap *map; + unsigned int dai_id = component->id; + + if (dai_id == LPASS_DP_RX) + map = drvdata->hdmiif_map; + else + map = drvdata->lpaif_map; + + regcache_cache_only(map, true); + regcache_mark_dirty(map); + + return 0; +} + +static int lpass_platform_pcmops_resume(struct snd_soc_component *component) +{ + struct lpass_data *drvdata = snd_soc_component_get_drvdata(component); + struct regmap *map; + unsigned int dai_id = component->id; + + if (dai_id == LPASS_DP_RX) + map = drvdata->hdmiif_map; + else + map = drvdata->lpaif_map; + + regcache_cache_only(map, false); + return regcache_sync(map); +} + + static const struct snd_soc_component_driver lpass_component_driver = { .name = DRV_NAME, .open = lpass_platform_pcmops_open, @@ -835,6 +868,8 @@ static const struct snd_soc_component_driver lpass_component_driver = { .mmap = lpass_platform_pcmops_mmap, .pcm_construct = lpass_platform_pcm_new, .pcm_destruct = lpass_platform_pcm_free, + .suspend = lpass_platform_pcmops_suspend, + .resume = lpass_platform_pcmops_resume, }; From 61c7dbec33777ade95d3db58beec8d7f177868c8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 17 Dec 2020 11:28:49 +0900 Subject: [PATCH 11/16] ASoC: rsnd: don't call clk_disable_unprepare() if can't use We need to care clock accessibility, because we might can't use clock for some reasons. It sets clk_rate for each clocks when enabled. This means it doesn't have clk_rate if we can't use. We can avoid to call clk_disable_unprepare() in such case. Link: https://lore.kernel.org/r/CAMuHMdWvB+p=2JqTsO7bR8uJqKqO5A2XgXFXsVAjHk3hcxgcTw@mail.gmail.com Reported-by: Geert Uytterhoeven Signed-off-by: Kuninori Morimoto Reviewed-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/87eejpgoi9.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index b9aacf3d3b29..abdfd9cf91e2 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -366,25 +366,27 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - int i, ret; + int i; for_each_rsnd_clk(clk, adg, i) { - ret = 0; if (enable) { - ret = clk_prepare_enable(clk); + int ret = clk_prepare_enable(clk); /* * We shouldn't use clk_get_rate() under * atomic context. Let's keep it when * rsnd_adg_clk_enable() was called */ - adg->clk_rate[i] = clk_get_rate(adg->clk[i]); + adg->clk_rate[i] = 0; + if (ret < 0) + dev_warn(dev, "can't use clk %d\n", i); + else + adg->clk_rate[i] = clk_get_rate(clk); } else { - clk_disable_unprepare(clk); + if (adg->clk_rate[i]) + clk_disable_unprepare(clk); + adg->clk_rate[i] = 0; } - - if (ret < 0) - dev_warn(dev, "can't use clk %d\n", i); } } From bb224c3e3e41d940612d4cc9573289cdbd5cb8f5 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Thu, 17 Dec 2020 11:54:01 +0100 Subject: [PATCH 12/16] ASoC: Intel: haswell: Add missing pm_ops haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20201217105401.27865-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c55d1239e705..c763bfeb1f38 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; From 6108f990c0887d3e8f1db2d13c7012e40a061f28 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Thu, 17 Dec 2020 16:56:51 +0800 Subject: [PATCH 13/16] ASoC: rt711: mutex between calibration and power state changes To avoid calibration time-out, this patch adds the mutex between calibration and power state changes Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20201217085651.24580-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5771c02c3459..85f744184a60 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -462,6 +462,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, unsigned int read_ll, read_rl; int i; + mutex_lock(&rt711->calibrate_mutex); + /* Can't use update bit function, so read the original value first */ addr_h = mc->reg; addr_l = mc->rreg; @@ -547,6 +549,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + mutex_unlock(&rt711->calibrate_mutex); return 0; } @@ -859,9 +863,11 @@ static int rt711_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: + mutex_lock(&rt711->calibrate_mutex); regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); break; default: From 349dd23931d1943b1083182e35715eba8b150fe1 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 17 Dec 2020 15:45:56 +0800 Subject: [PATCH 14/16] ASoC: max98373: don't access volatile registers in bias level off We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao Link: https://lore.kernel.org/r/20201217074556.32370-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-i2c.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373-sdw.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 8 ++++++++ 4 files changed, 79 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..85f6865019d4 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -19,6 +19,12 @@ #include #include "max98373.h" +static const u32 max98373_i2c_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R2000_SW_RESET, 0x00}, {MAX98373_R2001_INT_RAW1, 0x00}, @@ -472,6 +478,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -509,6 +520,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c, { int ret = 0; int reg = 0; + int i; struct max98373_priv *max98373 = NULL; max98373 = devm_kzalloc(&i2c->dev, sizeof(*max98373), GFP_KERNEL); @@ -534,6 +546,14 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache_num = ARRAY_SIZE(max98373_i2c_cache_reg); + max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_i2c_cache_reg[i]; + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..b8d471d79e93 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -23,6 +23,12 @@ struct sdw_stream_data { struct sdw_stream_runtime *sdw_stream; }; +static const u32 max98373_sdw_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R0040_SCP_INIT_STAT_1, 0x00}, {MAX98373_R0041_SCP_INIT_MASK_1, 0x00}, @@ -245,6 +251,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -757,6 +768,7 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) { struct max98373_priv *max98373; int ret; + int i; struct device *dev = &slave->dev; /* Allocate and assign private driver data structure */ @@ -768,6 +780,14 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache_num = ARRAY_SIZE(max98373_sdw_cache_reg); + max98373->cache = devm_kcalloc(dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_sdw_cache_reg[i]; + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..71f5a5228f34 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,11 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +217,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; From a84dfb3d55934253de6aed38ad75990278a2d21e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 17 Dec 2020 16:08:34 +0100 Subject: [PATCH 15/16] ASoC: meson: axg-tdmin: fix axg skew offset The signal captured on from tdm decoder of the AXG SoC is incorrect. It appears amplified. The skew offset of the decoder is wrong. Setting the skew offset to 3, like the g12 and sm1 SoCs, solves and gives correct data. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20201217150834.3247526-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdmin.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = { }; static const struct axg_tdm_formatter_driver axg_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); From 671ee4db952449acde126965bf76817a3159040d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 17 Dec 2020 16:08:12 +0100 Subject: [PATCH 16/16] ASoC: meson: axg-tdm-interface: fix loopback When the axg-tdm-interface was introduced, the backend DAI was marked as an endpoint when DPCM was walking the DAPM graph to find a its BE. It is no longer the case since this commit 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Because of this, when DPCM finds a BE it does everything it needs on the DAIs but it won't power up the widgets between the FE and the BE if there is no actual endpoint after the BE. On meson-axg HWs, the loopback is a special DAI of the tdm-interface BE. It is only linked to the dummy codec since there no actual HW after it. >From the DAPM perspective, the DAI has no endpoint. Because of this, the TDM decoder, which is a widget between the FE and BE is not powered up. >From the user perspective, everything seems fine but no data is produced. Connecting the Loopback DAI to a dummy DAPM endpoint solves the problem. Fixes: 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Cc: Charles Keepax Signed-off-by: Jerome Brunet Link: https://lore.kernel.org/r/20201217150812.3247405-1-jbrunet@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index c8664ab80d45..87cac440b369 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, return ret; } +static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = { + SND_SOC_DAPM_SIGGEN("Playback Signal"), +}; + +static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = { + { "Loopback", NULL, "Playback Signal" }, +}; + static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { - .set_bias_level = axg_tdm_iface_set_bias_level, + .dapm_widgets = axg_tdm_iface_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets), + .dapm_routes = axg_tdm_iface_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes), + .set_bias_level = axg_tdm_iface_set_bias_level, }; static const struct of_device_id axg_tdm_iface_of_match[] = {