From c20c6704bf2dafaba0d90c8310ef9e919fe4d2e2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 16 Nov 2017 04:36:51 +0000 Subject: ASoC: rcar: revert IOMMU support so far commit 4821d914fe74 ("ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU") had supported IOMMU, but it breaks normal sound "recorde" and both PulseAudio's "playback/recorde". The sound will be noisy. That commit was using dma_sync_single_for_xxx(), and driver should make sure memory is protected during CPU or Device are using it. But if driver returns current "residue" data size correctly on pointer function, player/recorder will access to protected memory. IOMMU feature should be supported, but I don't know how to handle it without memory cache problem at this point. Thus, this patch simply revert it to avoid current noisy sound. Tested-by: Hiroyuki Yokoyama Tested-by: Ryo Kodama Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 +-- sound/soc/sh/rcar/dma.c | 86 ++++-------------------------------------------- 2 files changed, 8 insertions(+), 82 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index c70eb2097816..f12a88a21dfa 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1332,8 +1332,8 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fd557abfe390..4d750bdf8e24 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -26,10 +26,7 @@ struct rsnd_dmaen { struct dma_chan *chan; dma_cookie_t cookie; - dma_addr_t dma_buf; unsigned int dma_len; - unsigned int dma_period; - unsigned int dma_cnt; }; struct rsnd_dmapp { @@ -71,38 +68,10 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -#define rsnd_dmaen_sync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 1) -#define rsnd_dmaen_unsync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 0) -static void __rsnd_dmaen_sync(struct rsnd_dmaen *dmaen, struct rsnd_dai_stream *io, - int i, int sync) -{ - struct device *dev = dmaen->chan->device->dev; - enum dma_data_direction dir; - int is_play = rsnd_io_is_play(io); - dma_addr_t buf; - int len, max; - size_t period; - - len = dmaen->dma_len; - period = dmaen->dma_period; - max = len / period; - i = i % max; - buf = dmaen->dma_buf + (period * i); - - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - - if (sync) - dma_sync_single_for_device(dev, buf, period, dir); - else - dma_sync_single_for_cpu(dev, buf, period, dir); -} - static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); bool elapsed = false; unsigned long flags; @@ -115,22 +84,9 @@ static void __rsnd_dmaen_complete(struct rsnd_mod *mod, */ spin_lock_irqsave(&priv->lock, flags); - if (rsnd_io_is_working(io)) { - rsnd_dmaen_unsync(dmaen, io, dmaen->dma_cnt); - - /* - * Next period is already started. - * Let's sync Next Next period - * see - * rsnd_dmaen_start() - */ - rsnd_dmaen_sync(dmaen, io, dmaen->dma_cnt + 2); - + if (rsnd_io_is_working(io)) elapsed = true; - dmaen->dma_cnt++; - } - spin_unlock_irqrestore(&priv->lock, flags); if (elapsed) @@ -165,14 +121,8 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - if (dmaen->chan) { - int is_play = rsnd_io_is_play(io); - + if (dmaen->chan) dmaengine_terminate_all(dmaen->chan); - dma_unmap_single(dmaen->chan->device->dev, - dmaen->dma_buf, dmaen->dma_len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - } return 0; } @@ -237,11 +187,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; - dma_addr_t buf; - size_t len; - size_t period; int is_play = rsnd_io_is_play(io); - int i; int ret; cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; @@ -258,19 +204,10 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - len = snd_pcm_lib_buffer_bytes(substream); - period = snd_pcm_lib_period_bytes(substream); - buf = dma_map_single(dmaen->chan->device->dev, - substream->runtime->dma_area, - len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - if (dma_mapping_error(dmaen->chan->device->dev, buf)) { - dev_err(dev, "dma map failed\n"); - return -EIO; - } - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - buf, len, period, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); @@ -282,18 +219,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, desc->callback = rsnd_dmaen_complete; desc->callback_param = rsnd_mod_get(dma); - dmaen->dma_buf = buf; - dmaen->dma_len = len; - dmaen->dma_period = period; - dmaen->dma_cnt = 0; - - /* - * synchronize this and next period - * see - * __rsnd_dmaen_complete() - */ - for (i = 0; i < 2; i++) - rsnd_dmaen_sync(dmaen, io, i); + dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); dmaen->cookie = dmaengine_submit(desc); if (dmaen->cookie < 0) { -- cgit v1.2.3-55-g7522 From 4c761ebfcb2d04ee36783c4c8c45ae00caf59d36 Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Fri, 3 Nov 2017 19:15:02 +0530 Subject: ASoC: Intel: kbl: Modify map for Headset Playback to fix pop-noise Patch fixes wrong path in commit 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop noise. Current topology for Headset results in unwanted pop noise, while switching from spk->hs at the start of Headset Playback. Hence re-introduced mixin-mixout dsp module in topology for headset playback pipe to fix the regression. And the corresponding modification for headset route is updated here. Fixes: 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") Signed-off-by: Naveen Manohar Signed-off-by: Sathya Prakash M R Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bcf20f3..6dcad0a8a0d0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164f2d43..271ae3c2c535 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, -- cgit v1.2.3-55-g7522 From bc6476d6c1edcb9b97621b5131bd169aa81f27db Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Mon, 13 Nov 2017 12:12:55 +0100 Subject: ASoC: da7218: fix fix child-node lookup Fix child-node lookup during probe, which ended up searching the whole device tree depth-first starting at the parent rather than just matching on its children. To make things worse, the parent codec node was also prematurely freed. Fixes: 4d50934abd22 ("ASoC: da7218: Add da7218 codec driver") Signed-off-by: Johan Hovold Acked-by: Adam Thomson Signed-off-by: Mark Brown Cc: stable --- sound/soc/codecs/da7218.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index b2d42ec1dcd9..56564ce90cb6 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2520,7 +2520,7 @@ static struct da7218_pdata *da7218_of_to_pdata(struct snd_soc_codec *codec) } if (da7218->dev_id == DA7218_DEV_ID) { - hpldet_np = of_find_node_by_name(np, "da7218_hpldet"); + hpldet_np = of_get_child_by_name(np, "da7218_hpldet"); if (!hpldet_np) return pdata; -- cgit v1.2.3-55-g7522 From 15f8c5f2415bfac73f33a14bcd83422bcbfb5298 Mon Sep 17 00:00:00 2001 From: Johan Hovold Date: Mon, 13 Nov 2017 12:12:56 +0100 Subject: ASoC: twl4030: fix child-node lookup Fix child-node lookup during probe, which ended up searching the whole device tree depth-first starting at the parent rather than just matching on its children. To make things worse, the parent codec node was also prematurely freed, while the child node was leaked. Fixes: 2d6d649a2e0f ("ASoC: twl4030: Support for DT booted kernel") Signed-off-by: Johan Hovold Signed-off-by: Mark Brown Cc: stable --- sound/soc/codecs/twl4030.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c482b2e7a7d2..cfe72b9d4356 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -232,7 +232,7 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); struct device_node *twl4030_codec_node = NULL; - twl4030_codec_node = of_find_node_by_name(codec->dev->parent->of_node, + twl4030_codec_node = of_get_child_by_name(codec->dev->parent->of_node, "codec"); if (!pdata && twl4030_codec_node) { @@ -241,9 +241,11 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) GFP_KERNEL); if (!pdata) { dev_err(codec->dev, "Can not allocate memory\n"); + of_node_put(twl4030_codec_node); return NULL; } twl4030_setup_pdata_of(pdata, twl4030_codec_node); + of_node_put(twl4030_codec_node); } return pdata; -- cgit v1.2.3-55-g7522 From 56986b07d17b4a19416e248aaca9367c241a824b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 22 Nov 2017 13:59:19 +0800 Subject: ASoC: rt5645: reset RT5645_AD_DA_MIXER at probe RT5645_AD_DA_MIXER (0x29) register will not be reset to default after SW reset. So we have to write it to its default value in i2c_probe. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 5f24df4fae8e..fcd02c2c76f1 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3823,6 +3823,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_read(regmap, RT5645_VENDOR_ID, &val); rt5645->v_id = val & 0xff; + regmap_write(rt5645->regmap, RT5645_AD_DA_MIXER, 0x8080); + ret = regmap_register_patch(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) -- cgit v1.2.3-55-g7522 From 254beff97b4714bac4ec8add5a6888c1adc1ad8f Mon Sep 17 00:00:00 2001 From: oder_chiou@realtek.com Date: Fri, 24 Nov 2017 16:11:22 +0800 Subject: ASoC: rt5514: Make sure the DMIC delay will be happened after normal SUPPLY widgets power on The patch makes sure the DMIC delay will be happened after normal SUPPLY widgets power on. If there are some platforms that provide the MCLK using the SUPPLY widget, it will make sure the delay time is helpful. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2a5b5d74e697..2dd6e9f990a4 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -496,7 +496,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, -- cgit v1.2.3-55-g7522 From fdaa451107ce543d345a339b4d5e20e8e4bac396 Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Mon, 20 Nov 2017 20:27:56 -0800 Subject: ASoC: amd: Add error checking to probe function The acp_audio_dma does not perform sufficient error checking in its probe function. This can result in crashes if a critical error path is encountered. Fixes: 7c31335a03b6a ("ASoC: AMD: add AMD ASoC ACP 2.x DMA driver") Cc: Alex Deucher Cc: Dominik Behr Cc: Daniel Kurtz Signed-off-by: Guenter Roeck Reviewed-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9f521a55d610..b5e41df6bb3a 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1051,6 +1051,11 @@ static int acp_audio_probe(struct platform_device *pdev) struct resource *res; const u32 *pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -ENODEV; + } + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); if (audio_drv_data == NULL) @@ -1058,6 +1063,8 @@ static int acp_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audio_drv_data->acp_mmio)) + return PTR_ERR(audio_drv_data->acp_mmio); /* The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' -- cgit v1.2.3-55-g7522 From 695b78b548d8a26288f041e907ff17758df9e1d5 Mon Sep 17 00:00:00 2001 From: Maciej S. Szmigiero Date: Mon, 20 Nov 2017 23:14:55 +0100 Subject: ASoC: fsl_ssi: AC'97 ops need regmap, clock and cleaning up on failure AC'97 ops (register read / write) need SSI regmap and clock, so they have to be set after them. We also need to set these ops back to NULL if we fail the probe. Signed-off-by: Maciej S. Szmigiero Acked-by: Nicolin Chen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/fsl/fsl_ssi.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f2f51e06e22c..c3a83ed0297e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1458,12 +1458,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) sizeof(fsl_ssi_ac97_dai)); fsl_ac97_data = ssi_private; - - ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); - if (ret) { - dev_err(&pdev->dev, "could not set AC'97 ops\n"); - return ret; - } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1574,6 +1568,14 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } + if (fsl_ssi_is_ac97(ssi_private)) { + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + goto error_ac97_ops; + } + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, &ssi_private->cpu_dai_drv, 1); if (ret) { @@ -1657,6 +1659,10 @@ error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); error_asoc_register: + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + +error_ac97_ops: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -- cgit v1.2.3-55-g7522 From b880b8056b31288323745a13930bc45cf4c86e9d Mon Sep 17 00:00:00 2001 From: Maciej S. Szmigiero Date: Mon, 20 Nov 2017 23:16:07 +0100 Subject: ASoC: fsl_ssi: serialize AC'97 register access operations AC'97 register access operations (both read and write) on SSI use a one, shared set of SSI registers for AC'97 register address and data. This means that only one such access is possible at a time and so all these operations need to be serialized. Since an AC'97 register access operation in this driver takes 100us+ let's use a mutex for this. Use this opportunity to also change a default value returned from AC'97 register read function from -1 to 0, since that's what AC'97 specs require to be returned when unknown / undefined registers are read. Signed-off-by: Maciej S. Szmigiero Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 26 ++++++++++++++++++++++---- 1 file changed, 22 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c3a83ed0297e..424bafaf51ef 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include #include #include +#include #include #include #include @@ -265,6 +266,8 @@ struct fsl_ssi_private { u32 fifo_watermark; u32 dma_maxburst; + + struct mutex ac97_reg_lock; }; /* @@ -1260,11 +1263,13 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, if (reg > 0x7f) return; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 write clk_prepare_enable failed: %d\n", ret); - return; + goto ret_unlock; } lreg = reg << 12; @@ -1278,6 +1283,9 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); clk_disable_unprepare(fsl_ac97_data->clk); + +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1285,16 +1293,18 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, { struct regmap *regs = fsl_ac97_data->regs; - unsigned short val = -1; + unsigned short val = 0; u32 reg_val; unsigned int lreg; int ret; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 read clk_prepare_enable failed: %d\n", ret); - return -1; + goto ret_unlock; } lreg = (reg & 0x7f) << 12; @@ -1309,6 +1319,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, clk_disable_unprepare(fsl_ac97_data->clk); +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); return val; } @@ -1569,6 +1581,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (fsl_ssi_is_ac97(ssi_private)) { + mutex_init(&ssi_private->ac97_reg_lock); ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); if (ret) { dev_err(&pdev->dev, "could not set AC'97 ops\n"); @@ -1663,6 +1676,9 @@ error_asoc_register: snd_soc_set_ac97_ops(NULL); error_ac97_ops: + if (fsl_ssi_is_ac97(ssi_private)) + mutex_destroy(&ssi_private->ac97_reg_lock); + if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); @@ -1681,8 +1697,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (fsl_ssi_is_ac97(ssi_private)) + if (fsl_ssi_is_ac97(ssi_private)) { snd_soc_set_ac97_ops(NULL); + mutex_destroy(&ssi_private->ac97_reg_lock); + } return 0; } -- cgit v1.2.3-55-g7522 From 346cccf88319344c9f513bd85df6ae2258e8a8ea Mon Sep 17 00:00:00 2001 From: oder_chiou@realtek.com Date: Mon, 20 Nov 2017 18:23:19 +0800 Subject: ASoC: rt5514: Add the sanity check for the driver_data in the resume function If the rt5514 spi driver is loaded, but the snd_soc_platform_driver is not loaded by the correct DAI settings, the NULL pointer will be gotten by snd_soc_platform_get_drvdata in the resume function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514-spi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 2df91db765ac..ca6a90d8fc39 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -482,10 +482,13 @@ static int __maybe_unused rt5514_resume(struct device *dev) if (device_may_wakeup(dev)) disable_irq_wake(irq); - if (rt5514_dsp->substream) { - rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf)); - if (buf[0] & RT5514_IRQ_STATUS_BIT) - rt5514_schedule_copy(rt5514_dsp); + if (rt5514_dsp) { + if (rt5514_dsp->substream) { + rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, + sizeof(buf)); + if (buf[0] & RT5514_IRQ_STATUS_BIT) + rt5514_schedule_copy(rt5514_dsp); + } } return 0; -- cgit v1.2.3-55-g7522 From a91d7fb97092d6b840af5899ded3b389603fd7f1 Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Tue, 28 Nov 2017 16:05:13 +0900 Subject: ASoC: rsnd: ssiu: clear SSI_MODE for non TDM Extended modes register SSI_MODE is set when SSI works in TDM Extended, but it isn't reset when SSI starts to work in other modes, thus causes issues. This patch clearss SSI_MODE register when SSI works in modes other than TDM Extended. Fixes: 186fadc132f0 ("ASoC: rsnd: add TDM Extend Mode support") Signed-off-by: Jiada Wang Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssiu.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d948757d300..6ff8a36c2c82 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -125,6 +125,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, { int hdmi = rsnd_ssi_hdmi_port(io); int ret; + u32 mode = 0; ret = rsnd_ssiu_init(mod, io, priv); if (ret < 0) @@ -136,9 +137,11 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, * see * rsnd_ssi_config_init() */ - rsnd_mod_write(mod, SSI_MODE, 0x1); + mode = 0x1; } + rsnd_mod_write(mod, SSI_MODE, mode); + if (rsnd_ssi_use_busif(io)) { rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr_bit(mod, io) | -- cgit v1.2.3-55-g7522 From 3c02a6d946657e1ae0688e0d89f2dd2cfe9afba8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Nov 2017 10:59:40 +0100 Subject: Revert "ALSA: usb-audio: Fix potential zero-division at parsing FU" The commit 8428a8ebde2d ("ALSA: usb-audio: Fix potential zero-division at parsing FU") is utterly bogus and breaks the case with csize=1 instead of fixing anything. Just take it back again. Reported-by: Jörg Otte Fixes: 8428a8ebde2d ("ALSA: usb-audio: Fix potential zero-division at parsing FU" Signed-off-by: Takashi Iwai Signed-off-by: Linus Torvalds --- sound/usb/mixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 0537c6322990..61b348383de8 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1476,9 +1476,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, return -EINVAL; } csize = hdr->bControlSize; - if (csize <= 1) { + if (!csize) { usb_audio_dbg(state->chip, - "unit %u: invalid bControlSize <= 1\n", + "unit %u: invalid bControlSize == 0\n", unitid); return -EINVAL; } -- cgit v1.2.3-55-g7522 From b89b6925bb9d48926d7ba713d3f13b14fc35c544 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 16 Nov 2017 11:55:18 -0800 Subject: ASoC: fsl_asrc: Fix typo in a field define ASRFSTi_IAEi has an 11-bit offset as its _SHIFT macro defines. So this patch just fixes that. Reported-by: Laurent Charpentier Signed-off-by: Nicolin Chen Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_asrc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..52c27a358933 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -260,8 +260,8 @@ #define ASRFSTi_OUTPUT_FIFO_SHIFT 12 #define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) #define ASRFSTi_IAEi_SHIFT 11 -#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) -#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_IAEi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_IAEi_SHIFT) #define ASRFSTi_INPUT_FIFO_WIDTH 7 #define ASRFSTi_INPUT_FIFO_SHIFT 0 #define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) -- cgit v1.2.3-55-g7522 From 43a3542870328601be02fcc9d27b09db467336ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Nov 2017 10:08:28 +0100 Subject: ALSA: seq: Remove spurious WARN_ON() at timer check The use of snd_BUG_ON() in ALSA sequencer timer may lead to a spurious WARN_ON() when a slave timer is deployed as its backend and a corresponding master timer stops meanwhile. The symptom was triggered by syzkaller spontaneously. Since the NULL timer is valid there, rip off snd_BUG_ON(). Reported-by: syzbot Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/seq_timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 37d9cfbc29f9..b80985fbc334 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -355,7 +355,7 @@ static int initialize_timer(struct snd_seq_timer *tmr) unsigned long freq; t = tmr->timeri->timer; - if (snd_BUG_ON(!t)) + if (!t) return -EINVAL; freq = tmr->preferred_resolution; -- cgit v1.2.3-55-g7522 From 51f493ae71adc2c49a317a13c38e54e1cdf46005 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 30 Nov 2017 10:15:02 +0000 Subject: ASoC: codecs: msm8916-wcd: Fix supported formats This codec is configurable for only 16 bit and 32 bit samples, so reflect this in the supported formats also remove 24bit sample from supported list. Signed-off-by: Srinivas Kandagatla Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 5f3c42c4f74a..066ea2f4ce7b 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -267,7 +267,7 @@ #define MSM8916_WCD_ANALOG_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4; diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index a10a724eb448..13354d6304a8 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -194,7 +194,7 @@ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_48000) #define MSM8916_WCD_DIGITAL_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) struct msm8916_wcd_digital_priv { struct clk *ahbclk, *mclk; @@ -645,7 +645,7 @@ static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, RX_I2S_CTL_RX_I2S_MODE_MASK, RX_I2S_CTL_RX_I2S_MODE_16); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: snd_soc_update_bits(dai->codec, LPASS_CDC_CLK_TX_I2S_CTL, TX_I2S_CTL_TX_I2S_MODE_MASK, TX_I2S_CTL_TX_I2S_MODE_32); -- cgit v1.2.3-55-g7522 From 737e0b7b67bdfe24090fab2852044bb283282fc5 Mon Sep 17 00:00:00 2001 From: Andrew F. Davis Date: Wed, 29 Nov 2017 15:32:46 -0600 Subject: ASoC: tlv320aic31xx: Fix GPIO1 register definition GPIO1 control register is number 51, fix this here. Fixes: bafcbfe429eb ("ASoC: tlv320aic31xx: Make the register values human readable") Signed-off-by: Andrew F. Davis Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic31xx.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 730fb2058869..1ff3edb7bbb6 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -116,7 +116,7 @@ struct aic31xx_pdata { /* INT2 interrupt control */ #define AIC31XX_INT2CTRL AIC31XX_REG(0, 49) /* GPIO1 control */ -#define AIC31XX_GPIO1 AIC31XX_REG(0, 50) +#define AIC31XX_GPIO1 AIC31XX_REG(0, 51) #define AIC31XX_DACPRB AIC31XX_REG(0, 60) /* ADC Instruction Set Register */ -- cgit v1.2.3-55-g7522 From 251552a2b0d454badc8f486e6d79100970c744b0 Mon Sep 17 00:00:00 2001 From: Jaejoong Kim Date: Mon, 4 Dec 2017 15:31:48 +0900 Subject: ALSA: usb-audio: Fix out-of-bound error The snd_usb_copy_string_desc() retrieves the usb string corresponding to the index number through the usb_string(). The problem is that the usb_string() returns the length of the string (>= 0) when successful, but it can also return a negative value about the error case or status of usb_control_msg(). If iClockSource is '0' as shown below, usb_string() will returns -EINVAL. This will result in '0' being inserted into buf[-22], and the following KASAN out-of-bound error message will be output. AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 1 bmAttributes 0x07 Internal programmable Clock (synced to SOF) bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 0 To fix it, check usb_string()'return value and bail out. ================================================================== BUG: KASAN: stack-out-of-bounds in parse_audio_unit+0x1327/0x1960 [snd_usb_audio] Write of size 1 at addr ffff88007e66735a by task systemd-udevd/18376 CPU: 0 PID: 18376 Comm: systemd-udevd Not tainted 4.13.0+ #3 Hardware name: LG Electronics 15N540-RFLGL/White Tip Mountain, BIOS 15N5 Call Trace: dump_stack+0x63/0x8d print_address_description+0x70/0x290 ? parse_audio_unit+0x1327/0x1960 [snd_usb_audio] kasan_report+0x265/0x350 __asan_store1+0x4a/0x50 parse_audio_unit+0x1327/0x1960 [snd_usb_audio] ? save_stack+0xb5/0xd0 ? save_stack_trace+0x1b/0x20 ? save_stack+0x46/0xd0 ? kasan_kmalloc+0xad/0xe0 ? kmem_cache_alloc_trace+0xff/0x230 ? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio] ? usb_audio_probe+0x4de/0xf40 [snd_usb_audio] ? usb_probe_interface+0x1f5/0x440 ? driver_probe_device+0x3ed/0x660 ? build_feature_ctl+0xb10/0xb10 [snd_usb_audio] ? save_stack_trace+0x1b/0x20 ? init_object+0x69/0xa0 ? snd_usb_find_csint_desc+0xa8/0xf0 [snd_usb_audio] snd_usb_mixer_controls+0x1dc/0x370 [snd_usb_audio] ? build_audio_procunit+0x890/0x890 [snd_usb_audio] ? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio] ? kmem_cache_alloc_trace+0xff/0x230 ? usb_ifnum_to_if+0xbd/0xf0 snd_usb_create_mixer+0x25b/0x4b0 [snd_usb_audio] ? snd_usb_create_stream+0x255/0x2c0 [snd_usb_audio] usb_audio_probe+0x4de/0xf40 [snd_usb_audio] ? snd_usb_autosuspend.part.7+0x30/0x30 [snd_usb_audio] ? __pm_runtime_idle+0x90/0x90 ? kernfs_activate+0xa6/0xc0 ? usb_match_one_id_intf+0xdc/0x130 ? __pm_runtime_set_status+0x2d4/0x450 usb_probe_interface+0x1f5/0x440 Cc: Signed-off-by: Jaejoong Kim Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 0537c6322990..8e18f7ec51f4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -204,6 +204,10 @@ static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen) { int len = usb_string(state->chip->dev, index, buf, maxlen - 1); + + if (len < 0) + return 0; + buf[len] = 0; return len; } -- cgit v1.2.3-55-g7522 From 89b89d121ffcf8d9546633b98ded9d18b8f75891 Mon Sep 17 00:00:00 2001 From: Jaejoong Kim Date: Mon, 4 Dec 2017 15:31:49 +0900 Subject: ALSA: usb-audio: Add check return value for usb_string() snd_usb_copy_string_desc() returns zero if usb_string() fails. In case of failure, we need to check the snd_usb_copy_string_desc()'s return value and add an exception case Signed-off-by: Jaejoong Kim Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 8e18f7ec51f4..afc208e1c756 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2178,13 +2178,14 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, if (len) ; else if (nameid) - snd_usb_copy_string_desc(state, nameid, kctl->id.name, + len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); - else { + else len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 0); - if (!len) - strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + + if (!len) { + strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); -- cgit v1.2.3-55-g7522 From f429e7e494afaded76e62c6f98211a635aa03098 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 5 Dec 2017 15:38:24 +0800 Subject: ALSA: hda/realtek - New codec support for ALC257 Add new support for ALC257 codec. [ It's supposed to be almost equivalent with other ALC25x variants, just adding another type and id -- tiwai ] Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 921a10eff43a..4b21f71d685c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -330,6 +330,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: @@ -2772,6 +2773,7 @@ enum { ALC269_TYPE_ALC298, ALC269_TYPE_ALC255, ALC269_TYPE_ALC256, + ALC269_TYPE_ALC257, ALC269_TYPE_ALC215, ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, @@ -2805,6 +2807,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC298: case ALC269_TYPE_ALC255: case ALC269_TYPE_ALC256: + case ALC269_TYPE_ALC257: case ALC269_TYPE_ALC215: case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: @@ -6867,6 +6870,10 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ break; + case 0x10ec0257: + spec->codec_variant = ALC269_TYPE_ALC257; + spec->gen.mixer_nid = 0; + break; case 0x10ec0215: case 0x10ec0285: case 0x10ec0289: @@ -7914,6 +7921,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260), HDA_CODEC_ENTRY(0x10ec0262, "ALC262", patch_alc262), HDA_CODEC_ENTRY(0x10ec0267, "ALC267", patch_alc268), -- cgit v1.2.3-55-g7522 From 362bca57f5d78220f8b5907b875961af9436e229 Mon Sep 17 00:00:00 2001 From: Robb Glasser Date: Tue, 5 Dec 2017 09:16:55 -0800 Subject: ALSA: pcm: prevent UAF in snd_pcm_info When the device descriptor is closed, the `substream->runtime` pointer is freed. But another thread may be in the ioctl handler, case SNDRV_CTL_IOCTL_PCM_INFO. This case calls snd_pcm_info_user() which calls snd_pcm_info() which accesses the now freed `substream->runtime`. Note: this fixes CVE-2017-0861 Signed-off-by: Robb Glasser Signed-off-by: Nick Desaulniers Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9070f277f8db..09ee8c6b9f75 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -153,7 +153,9 @@ static int snd_pcm_control_ioctl(struct snd_card *card, err = -ENXIO; goto _error; } + mutex_lock(&pcm->open_mutex); err = snd_pcm_info_user(substream, info); + mutex_unlock(&pcm->open_mutex); _error: mutex_unlock(®ister_mutex); return err; -- cgit v1.2.3-55-g7522 From c7b92172a61b91936be985cb9bc499a4ebc6489b Mon Sep 17 00:00:00 2001 From: Stefan Potyra Date: Wed, 6 Dec 2017 16:03:24 +0100 Subject: ASoC: rockchip: disable clock on error Disable the clocks in rk_spdif_probe when an error occurs after one of the clocks has been enabled previously. Found by Linux Driver Verification project (linuxtesting.org). Fixes: f874b80e1571 ASoC: rockchip: Add rockchip SPDIF transceiver driver Signed-off-by: Stefan Potyra Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_spdif.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ee5055d47d13..a89fe9b6463b 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -322,26 +322,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -373,6 +377,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } -- cgit v1.2.3-55-g7522 From e02b03303f13b6a571f01b4d84b69440696d2dde Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Wed, 6 Dec 2017 16:34:04 +0530 Subject: ASoC: Intel: Skylake: Do not check dev_type for dmic link type Some BIOS have inconsistent dev_type value for DMIC link type. Since there is only one device type for DMIC link type, remove device type check if link type is NHLT_LINK_DMIC. Signed-off-by: Guneshwor Singh Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-nhlt.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index d14c50a60289..3eaac41090ca 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -119,11 +119,16 @@ static bool skl_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt, if ((epnt->virtual_bus_id == instance_id) && (epnt->linktype == link_type) && - (epnt->direction == dirn) && - (epnt->device_type == dev_type)) - return true; - else - return false; + (epnt->direction == dirn)) { + /* do not check dev_type for DMIC link type */ + if (epnt->linktype == NHLT_LINK_DMIC) + return true; + + if (epnt->device_type == dev_type) + return true; + } + + return false; } struct nhlt_specific_cfg -- cgit v1.2.3-55-g7522 From 866f7ed7d67936dcdbcddc111c8af878c918fe7c Mon Sep 17 00:00:00 2001 From: Jussi Laako Date: Thu, 7 Dec 2017 12:58:33 +0200 Subject: ALSA: usb-audio: Add native DSD support for Esoteric D-05X Adds VID:PID of Esoteric D-05X to the TEAC device id's. Renames the is_teac_50X_dac() function to is_teac_dsd_dac() to cover broader device family from the same corporation sharing the same USB audio implementation. Signed-off-by: Jussi Laako Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 77eecaa4db1f..a66ef5777887 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1166,10 +1166,11 @@ static bool is_marantz_denon_dac(unsigned int id) /* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch * between PCM/DOP and native DSD mode */ -static bool is_teac_50X_dac(unsigned int id) +static bool is_teac_dsd_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ return true; } return false; @@ -1202,7 +1203,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_50X_dac(subs->stream->chip->usb_id)) { + } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1392,7 +1393,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* TEAC devices with USB DAC functionality */ - if (is_teac_50X_dac(chip->usb_id)) { + if (is_teac_dsd_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } -- cgit v1.2.3-55-g7522 From 2b4584d00a6bc02b63ab3c7213060d41a74bdff1 Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Thu, 7 Dec 2017 18:06:20 +0530 Subject: ALSA: hda - Add vendor id for Cannonlake HDMI codec Cannonlake HDMI codec has the same nid as Geminilake. This adds the codec entry for it. Signed-off-by: Guneshwor Singh Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c19c81d230bd..b4f1b6e88305 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -55,10 +55,11 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #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_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) || is_geminilake(codec) - + || is_kabylake(codec)) || is_geminilake(codec) \ + || is_cannonlake(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)) @@ -3841,6 +3842,7 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -- cgit v1.2.3-55-g7522 From 50dd2ea8ef67a1617e0c0658bcbec4b9fb03b936 Mon Sep 17 00:00:00 2001 From: Ben Hutchings Date: Fri, 8 Dec 2017 16:15:20 +0000 Subject: ASoC: wm_adsp: Fix validation of firmware and coeff lengths The checks for whether another region/block header could be present are subtracting the size from the current offset. Obviously we should instead subtract the offset from the size. The checks for whether the region/block data fit in the file are adding the data size to the current offset and header size, without checking for integer overflow. Rearrange these so that overflow is impossible. Signed-off-by: Ben Hutchings Acked-by: Charles Keepax Tested-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 65c059b5ffd7..66e32f5d2917 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1733,7 +1733,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) le64_to_cpu(footer->timestamp)); while (pos < firmware->size && - pos - firmware->size > sizeof(*region)) { + sizeof(*region) < firmware->size - pos) { region = (void *)&(firmware->data[pos]); region_name = "Unknown"; reg = 0; @@ -1782,8 +1782,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) regions, le32_to_cpu(region->len), offset, region_name); - if ((pos + le32_to_cpu(region->len) + sizeof(*region)) > - firmware->size) { + if (le32_to_cpu(region->len) > + firmware->size - pos - sizeof(*region)) { adsp_err(dsp, "%s.%d: %s region len %d bytes exceeds file length %zu\n", file, regions, region_name, @@ -2253,7 +2253,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && - pos - firmware->size > sizeof(*blk)) { + sizeof(*blk) < firmware->size - pos) { blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); @@ -2327,8 +2327,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > - firmware->size) { + if (le32_to_cpu(blk->len) > + firmware->size - pos - sizeof(*blk)) { adsp_err(dsp, "%s.%d: %s region len %d bytes exceeds file length %zu\n", file, blocks, region_name, -- cgit v1.2.3-55-g7522 From 0f0be40ba59c2d5fdfea48e3ff93f6165d616440 Mon Sep 17 00:00:00 2001 From: Alexandre Belloni Date: Fri, 8 Dec 2017 15:18:53 +0100 Subject: ASoC: atmel-classd: select correct Kconfig symbol SND_ATMEL_SOC_CLASSD selects SND_ATMEL_SOC_DMA but the driver itself handles its own DMA operations and doesn't need anything from atmel-pcm-dma.c or atmel_ssc_dai.c. Replace SND_ATMEL_SOC_DMA by SND_SOC_GENERIC_DMAENGINE_PCM which is the only one actually required. This may end up in a configuration leading to a link error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0x79c): undefined reference to `atmel_pcm_dma_platform_register' atmel_ssc_dai.c:(.text+0x79c): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `atmel_pcm_dma_platform_register' sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_put_audio': atmel_ssc_dai.c:(.text+0xf24): undefined reference to `atmel_pcm_dma_platform_unregister' atmel_ssc_dai.c:(.text+0xf24): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `atmel_pcm_dma_platform_unregister' Tested on sama5d2 xplained with the following configuration where nothing selects SND_ATMEL_SOC_DMA: CONFIG_SND_ATMEL_SOC=y CONFIG_SND_ATMEL_SOC_CLASSD=y Reported-by: Arnd Bergmann Tested-by: Arnd Bergmann Fixes: e0a25b6d1862 ("ASoC: atmel-classd: add the Audio Class D Amplifier") Signed-off-by: Alexandre Belloni Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4a56f3dfba51..dcee145dd179 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -64,7 +64,7 @@ config SND_AT91_SOC_SAM9X5_WM8731 config SND_ATMEL_SOC_CLASSD tristate "Atmel ASoC driver for boards using CLASSD" depends on ARCH_AT91 || COMPILE_TEST - select SND_ATMEL_SOC_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help Say Y if you want to add support for Atmel ASoC driver for boards using -- cgit v1.2.3-55-g7522 From 4362934a75ff2a399fd0bcd75937907115770020 Mon Sep 17 00:00:00 2001 From: Naveen Manohar Date: Fri, 8 Dec 2017 09:30:18 +0530 Subject: ASoC: Intel: Change kern log level to avoid unwanted messages patch suppresses the warning message "control load not supported" as this is a debug information to help debug issues in topology. Signed-off-by: Naveen Manohar Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a072bcf209d2..81923da18ac2 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2908,7 +2908,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, break; default: - dev_warn(bus->dev, "Control load not supported %d:%d:%d\n", + dev_dbg(bus->dev, "Control load not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); break; } -- cgit v1.2.3-55-g7522 From 33f801366bdf3f8b67dfe325b84f4051a090d01e Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Thu, 7 Dec 2017 22:15:38 -0800 Subject: ASoC: rsnd: ssi: fix race condition in rsnd_ssi_pointer_update Currently there is race condition between set of byte_pos and wrap it around when new buffer starts. If .pointer is called in-between it will result in inconsistent pointer position be returned from .pointer callback. This patch increments buffer pointer atomically to avoid this issue. Signed-off-by: Jiada Wang Reviewed-by: Takashi Sakamoto Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fece1e5f582f..cbf3bf312d23 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -446,25 +446,29 @@ static bool rsnd_ssi_pointer_update(struct rsnd_mod *mod, int byte) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + bool ret = false; + int byte_pos; - ssi->byte_pos += byte; + byte_pos = ssi->byte_pos + byte; - if (ssi->byte_pos >= ssi->next_period_byte) { + if (byte_pos >= ssi->next_period_byte) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); ssi->period_pos++; ssi->next_period_byte += ssi->byte_per_period; if (ssi->period_pos >= runtime->periods) { - ssi->byte_pos = 0; + byte_pos = 0; ssi->period_pos = 0; ssi->next_period_byte = ssi->byte_per_period; } - return true; + ret = true; } - return false; + WRITE_ONCE(ssi->byte_pos, byte_pos); + + return ret; } /* @@ -838,7 +842,7 @@ static int rsnd_ssi_pointer(struct rsnd_mod *mod, struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - *pointer = bytes_to_frames(runtime, ssi->byte_pos); + *pointer = bytes_to_frames(runtime, READ_ONCE(ssi->byte_pos)); return 0; } -- cgit v1.2.3-55-g7522 From 958d022e326810fd762505bd02007aced79ffcbc Mon Sep 17 00:00:00 2001 From: oder_chiou@realtek.com Date: Thu, 14 Dec 2017 09:54:07 +0800 Subject: ASoC: rt5663: Fix the wrong result of the first jack detection In the first jack detection while booting, the result will always show as headset, even we insert the headphone. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5663.c | 4 ++++ sound/soc/codecs/rt5663.h | 4 ++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index b036c9dc0c8c..d329bf719d80 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1560,6 +1560,10 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) RT5663_IRQ_POW_SAV_MASK, RT5663_IRQ_POW_SAV_EN); snd_soc_update_bits(codec, RT5663_IRQ_1, RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_EM_JD_MASK, RT5663_EM_JD_RST); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_EM_JD_MASK, RT5663_EM_JD_NOR); while (true) { regmap_read(rt5663->regmap, RT5663_INT_ST_2, &val); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index c5a9b69579ad..03adc8004ba9 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -1029,6 +1029,10 @@ #define RT5663_POL_EXT_JD_SHIFT 10 #define RT5663_POL_EXT_JD_EN (0x1 << 10) #define RT5663_POL_EXT_JD_DIS (0x0 << 10) +#define RT5663_EM_JD_MASK (0x1 << 7) +#define RT5663_EM_JD_SHIFT 7 +#define RT5663_EM_JD_NOR (0x1 << 7) +#define RT5663_EM_JD_RST (0x0 << 7) /* DACREF LDO Control (0x0112)*/ #define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) -- cgit v1.2.3-55-g7522 From c1cfd9025cc394fd137a01159d74335c5ac978ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Dec 2017 16:44:12 +0100 Subject: ALSA: rawmidi: Avoid racy info ioctl via ctl device The rawmidi also allows to obtaining the information via ioctl of ctl API. It means that user can issue an ioctl to the rawmidi device even when it's being removed as long as the control device is present. Although the code has some protection via the global register_mutex, its range is limited to the search of the corresponding rawmidi object, and the mutex is already unlocked at accessing the rawmidi object. This may lead to a use-after-free. For avoiding it, this patch widens the application of register_mutex to the whole snd_rawmidi_info_select() function. We have another mutex per rawmidi object, but this operation isn't very hot path, so it shouldn't matter from the performance POV. Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b3b353d72527..f055ca10bbc1 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -579,15 +579,14 @@ static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream, return 0; } -int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +static int __snd_rawmidi_info_select(struct snd_card *card, + struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; struct snd_rawmidi_str *pstr; struct snd_rawmidi_substream *substream; - mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, info->device); - mutex_unlock(®ister_mutex); if (!rmidi) return -ENXIO; if (info->stream < 0 || info->stream > 1) @@ -603,6 +602,16 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info } return -ENXIO; } + +int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +{ + int ret; + + mutex_lock(®ister_mutex); + ret = __snd_rawmidi_info_select(card, info); + mutex_unlock(®ister_mutex); + return ret; +} EXPORT_SYMBOL(snd_rawmidi_info_select); static int snd_rawmidi_info_select_user(struct snd_card *card, -- cgit v1.2.3-55-g7522 From 9226665159f0367ad08bc7d5dd194aeadb90316f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 14 Dec 2017 15:28:58 +0800 Subject: ALSA: hda/realtek - Fix Dell AIO LineOut issue Dell AIO had LineOut jack. Add LineOut verb into this patch. [ Additional notes: the ALC274 codec seems requiring the fixed pin / DAC connections for HP / line-out pins for enabling EQ for speakers; i.e. the HP / LO pins expect to be connected with NID 0x03 while keeping the speaker with NID 0x02. However, by adding a new line-out pin, the auto-parser assigns the NID 0x02 for HP/LO pins as primary outputs. As an easy workaround, we provide the preferred_pairs[] to map forcibly for these pins. -- tiwai ] Fixes: 75ee94b20b46 ("ALSA: hda - fix headset mic problem for Dell machines with alc274") Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4b21f71d685c..6a4db00511ab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5185,6 +5185,22 @@ static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, } } +/* Forcibly assign NID 0x03 to HP/LO while NID 0x02 to SPK for EQ */ +static void alc274_fixup_bind_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t preferred_pairs[] = { + 0x21, 0x03, 0x1b, 0x03, 0x16, 0x02, + 0 + }; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.preferred_dacs = preferred_pairs; +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5302,6 +5318,8 @@ enum { ALC233_FIXUP_LENOVO_MULTI_CODECS, ALC294_FIXUP_LENOVO_MIC_LOCATION, ALC700_FIXUP_INTEL_REFERENCE, + ALC274_FIXUP_DELL_BIND_DACS, + ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, }; static const struct hda_fixup alc269_fixups[] = { @@ -6112,6 +6130,21 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC274_FIXUP_DELL_BIND_DACS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc274_fixup_bind_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, + [ALC274_FIXUP_DELL_AIO_LINEOUT_VERB] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x0401102f }, + { } + }, + .chained = true, + .chain_id = ALC274_FIXUP_DELL_BIND_DACS + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6578,7 +6611,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x12, 0xb7a60130}, {0x13, 0xb8a61140}, {0x16, 0x90170110}, -- cgit v1.2.3-55-g7522 From 5a15f289ee87eaf33f13f08a4909ec99d837ec5f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Dec 2017 23:36:57 +0100 Subject: ALSA: usb-audio: Fix the missing ctl name suffix at parsing SU The commit 89b89d121ffc ("ALSA: usb-audio: Add check return value for usb_string()") added the check of the return value from snd_usb_copy_string_desc(), which is correct per se, but it introduced a regression. In the original code, either the "Clock Source", "Playback Source" or "Capture Source" suffix is added after the terminal string, while the commit changed it to add the suffix only when get_term_name() is failing. It ended up with an incorrect ctl name like "PCM" instead of "PCM Capture Source". Also, even the original code has a similar bug: when the ctl name is generated from snd_usb_copy_string_desc() for the given iSelector, it also doesn't put the suffix. This patch addresses these issues: the suffix is added always when no static mapping is found. Also the patch tries to put more comments and cleans up the if/else block for better readability in order to avoid the same pitfall again. Fixes: 89b89d121ffc ("ALSA: usb-audio: Add check return value for usb_string()") Reported-and-tested-by: Mauro Santos Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 27 ++++++++++++++++----------- 1 file changed, 16 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index afc208e1c756..60ebc99ae323 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2173,20 +2173,25 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; - nameid = uac_selector_unit_iSelector(desc); + /* check the static mapping table at first */ len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); - if (len) - ; - else if (nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, - sizeof(kctl->id.name)); - else - len = get_term_name(state, &state->oterm, - kctl->id.name, sizeof(kctl->id.name), 0); - if (!len) { - strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* no mapping ? */ + /* if iSelector is given, use it */ + nameid = uac_selector_unit_iSelector(desc); + if (nameid) + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, + sizeof(kctl->id.name)); + /* ... or pick up the terminal name at next */ + if (!len) + len = get_term_name(state, &state->oterm, + kctl->id.name, sizeof(kctl->id.name), 0); + /* ... or use the fixed string "USB" as the last resort */ + if (!len) + strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* and add the proper suffix */ if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); else if ((state->oterm.type & 0xff00) == 0x0100) -- cgit v1.2.3-55-g7522 From d070f7c703ef26e3db613f24206823f916272fc6 Mon Sep 17 00:00:00 2001 From: Abhijeet Kumar Date: Tue, 12 Dec 2017 00:40:25 +0530 Subject: ASoC: nau8825: fix issue that pop noise when start capture In skylake platform, we hear a loud pop noise(0 dB) at start of audio capture power up sequence. This patch removes the pop noise from the recording by adding a delay before enabling ADC. Signed-off-by: Abhijeet Kumar Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 714ce17da717..e853a6dfd33b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -905,6 +905,7 @@ static int nau8825_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + msleep(125); regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); break; -- cgit v1.2.3-55-g7522 From 20220945b1a8e77c789dd4bb9aa1471b6e8695cc Mon Sep 17 00:00:00 2001 From: Brian Norris Date: Fri, 15 Dec 2017 20:07:23 -0800 Subject: ASoC: rt5514-spi: only enable wakeup when fully initialized If an rt5514-spi device is probed but the platform hasn't linked it in, we might never fully request the SPI IRQ, nor configure the rt5514 DSP, but we still might try to enable the SPI IRQ (enable_irq_wake()). This is bad, and among other things, can cause the interrupt to trigger every time we try to suspend the system (e.g., because the interrupt trigger setting was never set properly). Instead of setting our wakeup capabilities in the SPI driver probe routine, let's wait until we've actually requested the IRQ. Fixes issues seen on the "kevin" Chromebook (Samsung Chromebook Plus). Fixes: 58f1c07d23cd ("ASoC: rt5514: Voice wakeup support.") Signed-off-by: Brian Norris Signed-off-by: Mark Brown --- sound/soc/codecs/rt5514-spi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index ca6a90d8fc39..64bf26cec20d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -289,6 +289,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) dev_err(&rt5514_spi->dev, "%s Failed to reguest IRQ: %d\n", __func__, ret); + else + device_init_wakeup(rt5514_dsp->dev, true); } return 0; @@ -456,8 +458,6 @@ static int rt5514_spi_probe(struct spi_device *spi) return ret; } - device_init_wakeup(&spi->dev, true); - return 0; } -- cgit v1.2.3-55-g7522 From d5aa24825da5711f8cb829f873160ddf1a29b19c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 20 Dec 2017 06:11:59 +0000 Subject: ASoC: rsnd: fixup ADG register mask BRGCKR should use 0x80770000, instead of 0x80FF0000. R-Car Gen2 xxx_TIMSEL should use 0x0F1F, R-Car Gen3 xxx_TIMSEL should use 0x1F1F. Here, Gen3 doesn't support AVD, thus, both case can use 0x0F1F. Signed-off-by: Kuninori Morimoto Reviewed-by: Hiroyuki Yokoyama Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 8ddb08714faa..4672688cac32 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -222,7 +222,7 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *cmd_mod, NULL, &val, NULL); val = val << shift; - mask = 0xffff << shift; + mask = 0x0f1f << shift; rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val); @@ -250,7 +250,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, in = in << shift; out = out << shift; - mask = 0xffff << shift; + mask = 0x0f1f << shift; switch (id / 2) { case 0: @@ -380,7 +380,7 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) ckr = 0x80000000; } - rsnd_mod_bset(adg_mod, BRGCKR, 0x80FF0000, adg->ckr | ckr); + rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr | ckr); rsnd_mod_write(adg_mod, BRRA, adg->rbga); rsnd_mod_write(adg_mod, BRRB, adg->rbgb); -- cgit v1.2.3-55-g7522 From 322f74ede933b3e2cb78768b6a6fdbfbf478a0c1 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 22 Dec 2017 11:17:44 +0800 Subject: ALSA: hda - Add MIC_NO_PRESENCE fixup for 2 HP machines There is a headset jack on the front panel, when we plug a headset into it, the headset mic can't trigger unsol events, and read_pin_sense() can't detect its presence too. So add this fixup to fix this issue. Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a81aacf684b2..37e1cf8218ff 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -271,6 +271,8 @@ enum { CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, CXT_FIXUP_MUTE_LED_GPIO, + CXT_FIXUP_HEADSET_MIC, + CXT_FIXUP_HP_MIC_NO_PRESENCE, }; /* for hda_fixup_thinkpad_acpi() */ @@ -350,6 +352,18 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, } } +static void cxt_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + break; + } +} + /* OPLC XO 1.5 fixup */ /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors) @@ -880,6 +894,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_gpio, }, + [CXT_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headset_mic, + }, + [CXT_FIXUP_HP_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02a1113c }, + { } + }, + .chained = true, + .chain_id = CXT_FIXUP_HEADSET_MIC, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -934,6 +961,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", 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(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- cgit v1.2.3-55-g7522 From 285d5ddcffafa5d5e68c586f4c9eaa8b24a2897d Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 22 Dec 2017 11:17:45 +0800 Subject: ALSA: hda - fix headset mic detection issue on a Dell machine It has the codec alc256, and add its pin definition to pin quirk table to let it apply ALC255_FIXUP_DELL1_MIC_NO_PRESENCE. Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a4db00511ab..682858548b9b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6585,6 +6585,11 @@ 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}, -- cgit v1.2.3-55-g7522 From 8da5bbfc7cbba909f4f32d5e1dda3750baa5d853 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 22 Dec 2017 11:17:46 +0800 Subject: ALSA: hda - change the location for one mic on a Lenovo machine There are two front mics on this machine, and current driver assign the same name Mic to both of them, but pulseaudio can't handle them. As a workaround, we change the location for one of them, then the driver will assign "Front Mic" and "Mic" for them. Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 682858548b9b..1522ba31e16d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6328,6 +6328,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), -- cgit v1.2.3-55-g7522 From a36c2638380c0a4676647a1f553b70b20d3ebce1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Dec 2017 10:45:07 +0100 Subject: ALSA: hda: Drop useless WARN_ON() Since the commit 97cc2ed27e5a ("ALSA: hda - Fix yet another i915 pointer leftover in error path") cleared hdac_acomp pointer, the WARN_ON() non-NULL check in snd_hdac_i915_register_notifier() may give a false-positive warning, as the function gets called no matter whether the component is registered or not. For fixing it, let's get rid of the spurious WARN_ON(). Fixes: 97cc2ed27e5a ("ALSA: hda - Fix yet another i915 pointer leftover in error path") Cc: Reported-by: Kouta Okamoto Signed-off-by: Takashi Iwai --- sound/hda/hdac_i915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 038a180d3f81..cbe818eda336 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -325,7 +325,7 @@ static int hdac_component_master_match(struct device *dev, void *data) */ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) { - if (WARN_ON(!hdac_acomp)) + if (!hdac_acomp) return -ENODEV; hdac_acomp->audio_ops = aops; -- cgit v1.2.3-55-g7522 From 44be77c590f381bc629815ac789b8b15ecc4ddcf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Dec 2017 08:53:59 +0100 Subject: ALSA: hda - Fix missing COEF init for ALC225/295/299 There was a long-standing problem on HP Spectre X360 with Kabylake where it lacks of the front speaker output in some situations. Also there are other products showing the similar behavior. The culprit seems to be the missing COEF setup on ALC codecs, ALC225/295/299, which are all compatible. This patch adds the proper COEF setup (to initialize idx 0x67 / bits 0x3000) for addressing the issue. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195457 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1522ba31e16d..8fd2d9c62c96 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -324,8 +324,12 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0292: alc_update_coef_idx(codec, 0x4, 1<<15, 0); break; - case 0x10ec0215: case 0x10ec0225: + case 0x10ec0295: + case 0x10ec0299: + alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); + /* fallthrough */ + case 0x10ec0215: case 0x10ec0233: case 0x10ec0236: case 0x10ec0255: @@ -336,10 +340,8 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0286: case 0x10ec0288: case 0x10ec0285: - case 0x10ec0295: case 0x10ec0298: case 0x10ec0289: - case 0x10ec0299: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; case 0x10ec0275: -- cgit v1.2.3-55-g7522