From 6fce983f9b3ef51d47e647b2cff15049ef803781 Mon Sep 17 00:00:00 2001 From: Jose Abreu Date: Tue, 13 Dec 2016 11:03:49 +0000 Subject: [PATCH 01/17] ASoC: dwc: Fix PIO mode initialization We can no longer rely on the return value of devm_snd_dmaengine_pcm_register(...) to check if the DMA handle is declared in the DT. Previously this check activated PIO mode but currently dma_request_chan returns either a valid channel or -EPROBE_DEFER. In order to activate PIO mode check instead if the interrupt line is declared. This reflects better what is documented in the DT bindings (see Documentation/devicetree/bindings/sound/ designware-i2s.txt). Also, initialize use_pio variable which was never being set causing PIO mode to never work. Signed-off-by: Jose Abreu Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 2998954a1c74..bdf8398cbc81 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -681,22 +681,19 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (!pdata) { - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); - if (ret == -EPROBE_DEFER) { - dev_err(&pdev->dev, - "failed to register PCM, deferring probe\n"); - return ret; - } else if (ret) { - dev_err(&pdev->dev, - "Could not register DMA PCM: %d\n" - "falling back to PIO mode\n", ret); + if (irq >= 0) { ret = dw_pcm_register(pdev); - if (ret) { - dev_err(&pdev->dev, - "Could not register PIO PCM: %d\n", + dev->use_pio = true; + } else { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + 0); + dev->use_pio = false; + } + + if (ret) { + dev_err(&pdev->dev, "could not register pcm: %d\n", ret); - goto err_clk_disable; - } + goto err_clk_disable; } } From 0ea617a298dcdc2251b4e10f83ac3f3e627b66e3 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 8 Dec 2016 13:05:43 +0000 Subject: [PATCH 02/17] ASoC: rsnd: don't double free kctrl On an error, snd_ctl_add already free's kctrl, so calling snd_ctl_free_one to free it again leads to a double free error. Fix this by removing the extraneous snd_ctl_free_one call. Issue found using static analysis with CoverityScan, CID 1372908 Signed-off-by: Colin Ian King Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4bd68de76130..99b5b0835c1e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1030,10 +1030,8 @@ static int __rsnd_kctrl_new(struct rsnd_mod *mod, return -ENOMEM; ret = snd_ctl_add(card, kctrl); - if (ret < 0) { - snd_ctl_free_one(kctrl); + if (ret < 0) return ret; - } cfg->update = update; cfg->card = card; From c2b36129ce53a22b89dd2b88db33e7ffdefe0f41 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Fri, 9 Dec 2016 14:17:47 +0000 Subject: [PATCH 03/17] ASoC: topology: kfree kcontrol->private_value before freeing kcontrol kcontrol->private_value is being kfree'd after kcontrol has been freed (in previous call to snd_ctl_remove). Instead, fix this by kfreeing the private_value before kcontrol. CoverityScan CID#1388311 "Read from pointer after free" Fixes: eea3dd4f1247a ("ASoC: topology: Only free TLV for volume mixers of a widget") Signed-off-by: Colin Ian King Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 65670b2b408c..fbfb1fab88d5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -514,13 +514,12 @@ static void remove_widget(struct snd_soc_component *comp, == SND_SOC_TPLG_TYPE_MIXER) kfree(kcontrol->tlv.p); - snd_ctl_remove(card, kcontrol); - /* Private value is used as struct soc_mixer_control * for volume mixers or soc_bytes_ext for bytes * controls. */ kfree((void *)kcontrol->private_value); + snd_ctl_remove(card, kcontrol); } kfree(w->kcontrol_news); } From 9e4d59ada4d602e78eee9fb5f898ce61fdddb446 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 16 Dec 2016 18:26:54 +0900 Subject: [PATCH 04/17] ASoC: hdmi-codec: use unsigned type to structure members with bit-field This is a fix for Linux 4.10-rc1. In C language specification, a bit-field is interpreted as a signed or unsigned integer type consisting of the specified number of bits. In GCC manual, the range of a signed bit field of N bits is from -(2^N) / 2 to ((2^N) / 2) - 1 https://www.gnu.org/software/gnu-c-manual/gnu-c-manual.html#Bit-Fields Therefore, when defined as 1 bit-field with signed type, variables can represents -1 and 0. The snd-soc-hdmi-codec module includes a structure which has signed type members with bit-fields. Codes of this module assign 0 and 1 to the members. This seems to result in implementation-dependent behaviours. As of v4.10-rc1 merge window, outside of sound subsystem, this structure is referred by below GPU modules. - tda998x - sti-drm - mediatek-drm-hdmi - msm As long as I review their codes relevant to the structure, the structure members are used just for condition statements and printk formats. My proposal of change is a bit intrusive to the printk formats but this may be acceptable. Totally, it's reasonable to use unsigned type for the structure members. This bug is detected by Sparse, static code analyzer with below warnings. ./include/sound/hdmi-codec.h:39:26: error: dubious one-bit signed bitfield ./include/sound/hdmi-codec.h:40:28: error: dubious one-bit signed bitfield ./include/sound/hdmi-codec.h:41:29: error: dubious one-bit signed bitfield ./include/sound/hdmi-codec.h:42:31: error: dubious one-bit signed bitfield Fixes: 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Signed-off-by: Takashi Sakamoto Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown CC: stable@vger.kernel.org --- include/sound/hdmi-codec.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 530c57bdefa0..915c4357945c 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -36,10 +36,10 @@ struct hdmi_codec_daifmt { HDMI_AC97, HDMI_SPDIF, } fmt; - int bit_clk_inv:1; - int frame_clk_inv:1; - int bit_clk_master:1; - int frame_clk_master:1; + unsigned int bit_clk_inv:1; + unsigned int frame_clk_inv:1; + unsigned int bit_clk_master:1; + unsigned int frame_clk_master:1; }; /* From 4a8b3a682be9addff7dbd16371fa8c34103b5c31 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 16 Dec 2016 10:55:49 -0600 Subject: [PATCH 05/17] ASoC: Intel: bytcr_rt5640: fallback mechanism if MCLK is not enabled Commit df1a2776a795 ("ASoC: Intel: bytcr_rt5640: add MCLK support") was merged but the corresponding clock framework patches have not, after being bumped from audio to clock to x86 domains. The missing clock-related patches result in a regression starting with 4.9 with the audio card not being created. Rather than reverting this commit and all following updates already queued up for 4.10, handle run-time dependency on MCLK and fall back to the previous bit-clock mode. This provides the same functionality as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain silent with this fallback but additional patches are needed anyway. As suggested by Mark Brown, the fallback is only allowed with -ENOENT, all other run-time errors, including -EPROBE_DEFER, will stop the probe with no sound card registered. This patch should be applied to -stable as well as ASoC 4.10 fixes Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 507a86a5eafe..e33e4777a65c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -825,10 +825,20 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && (is_valleyview())) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); + dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); - return PTR_ERR(priv->mclk); + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret_val); + + /* + * Fall back to bit clock usage for -ENOENT (clock not + * available likely due to missing dependencies), bail + * for all other errors, including -EPROBE_DEFER + */ + if (ret_val != -ENOENT) + return ret_val; + byt_rt5640_quirk &= ~BYT_RT5640_MCLK_EN; } } From 1cab2a84f470e15ecc8e5143bfe9398c6e888032 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 20 Dec 2016 10:29:12 +0000 Subject: [PATCH 06/17] ASoC: wm_adsp: Don't overrun firmware file buffer when reading region data Protect against corrupt firmware files by ensuring that the length we get for the data in a region actually lies within the available firmware file data buffer. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 593b7d1aed46..d72ccef9e238 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1551,7 +1551,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wmfw_region *region; const struct wm_adsp_region *mem; const char *region_name; - char *file, *text; + char *file, *text = NULL; struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; @@ -1700,10 +1700,21 @@ 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) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %zu\n", + file, regions, region_name, + le32_to_cpu(region->len), firmware->size); + ret = -EINVAL; + goto out_fw; + } + if (text) { memcpy(text, region->data, le32_to_cpu(region->len)); adsp_info(dsp, "%s: %s\n", file, text); kfree(text); + text = NULL; } if (reg) { @@ -1748,6 +1759,7 @@ out_fw: regmap_async_complete(regmap); wm_adsp_buf_free(&buf_list); release_firmware(firmware); + kfree(text); out: kfree(file); @@ -2233,6 +2245,17 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %zu\n", + file, blocks, region_name, + le32_to_cpu(blk->len), + firmware->size); + ret = -EINVAL; + goto out_fw; + } + buf = wm_adsp_buf_alloc(blk->data, le32_to_cpu(blk->len), &buf_list); From d2e3a1358c37cd82eef92b5e908b4f0472194481 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Thu, 29 Dec 2016 14:11:21 +0100 Subject: [PATCH 07/17] ASoC: Fix binding and probing of auxiliary components Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 10 +++++----- 2 files changed, 8 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2b502f6cc6d0..b86168a21d56 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -813,6 +813,7 @@ struct snd_soc_component { unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; + struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1152,6 +1153,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1547,6 +1549,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f1901bb1466e..baa1afa41e3d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1748,6 +1748,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) component->init = aux_dev->init; component->auxiliary = 1; + list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1758,16 +1759,14 @@ err_defer: static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp; + struct snd_soc_component *comp, *tmp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, + card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1776,6 +1775,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } + list_del(&comp->card_aux_list); } } } From 63c3194b82530bd71fd49db84eb7ab656b8d404a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 23 Dec 2016 11:21:10 +0200 Subject: [PATCH 08/17] ASoC: tlv320aic3x: Mark the RESET register as volatile The RESET register only have one self clearing bit and it should not be cached. If it is cached, when we sync the registers back to the chip we will initiate a software reset as well, which is not desirable. Signed-off-by: Peter Ujfalusi Reviewed-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8877b74b0510..bb94d50052d7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -126,6 +126,16 @@ static const struct reg_default aic3x_reg[] = { { 108, 0x00 }, { 109, 0x00 }, }; +static bool aic3x_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC3X_RESET: + return true; + default: + return false; + } +} + static const struct regmap_config aic3x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -133,6 +143,9 @@ static const struct regmap_config aic3x_regmap = { .max_register = DAC_ICC_ADJ, .reg_defaults = aic3x_reg, .num_reg_defaults = ARRAY_SIZE(aic3x_reg), + + .volatile_reg = aic3x_volatile_reg, + .cache_type = REGCACHE_RBTREE, }; From 91ce54978ccece323aa6df930249ff84a7d233c7 Mon Sep 17 00:00:00 2001 From: G Kranthi Date: Tue, 20 Dec 2016 12:46:45 +0530 Subject: [PATCH 09/17] ASoC: Intel: Skylake: Fix to fail safely if module not available in path If a module is not available in a pipeline, fail safely rather than causing oops. Signed-off-by: G Kranthi Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 84b5101e6ca6..6c6b63a6b338 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -180,6 +180,9 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); + if (!mconfig) + return -EINVAL; + skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); return 0; From a33b56a6a824fa5cd89c74f85cbeb9af1dcef87e Mon Sep 17 00:00:00 2001 From: John Hsu Date: Tue, 20 Dec 2016 16:47:06 +0800 Subject: [PATCH 10/17] ASoC: nau8825: correct the function name of register Change to correct name of the register function. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 6 +++--- sound/soc/codecs/nau8825.h | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index efe3a44658d5..abf77dd422f4 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -561,9 +561,9 @@ static void nau8825_xtalk_prepare(struct nau8825 *nau8825) nau8825_xtalk_backup(nau8825); /* Config IIS as master to output signal by codec */ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_MASTER | - (0x2 << NAU8825_I2S_DRV_SFT) | 0x1); + (0x2 << NAU8825_I2S_LRC_DIV_SFT) | 0x1); /* Ramp up headphone volume to 0dB to get better performance and * avoid pop noise in headphone. */ @@ -657,7 +657,7 @@ static void nau8825_xtalk_clean(struct nau8825 *nau8825) NAU8825_IRQ_RMS_EN, NAU8825_IRQ_RMS_EN); /* Recover default value for IIS */ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_SLAVE); /* Restore value of specific register for cross talk */ nau8825_xtalk_restore(nau8825); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5d1704e73241..b6b21b312854 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -247,8 +247,8 @@ /* I2S_PCM_CTRL2 (0x1d) */ #define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ -#define NAU8825_I2S_DRV_SFT 12 -#define NAU8825_I2S_DRV_MASK (0x3 << NAU8825_I2S_DRV_SFT) +#define NAU8825_I2S_LRC_DIV_SFT 12 +#define NAU8825_I2S_LRC_DIV_MASK (0x3 << NAU8825_I2S_LRC_DIV_SFT) #define NAU8825_I2S_MS_SFT 3 #define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) From a1792cda51300e15b03549cccf0b09f3be82e697 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Tue, 20 Dec 2016 12:03:09 +0800 Subject: [PATCH 11/17] ASoC: nau8825: fix invalid configuration in Pre-Scalar of FLL The clk_ref_div is not configured in the correct position of the register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned the wrong value. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 3 ++- sound/soc/codecs/nau8825.h | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index abf77dd422f4..4576f987a4a5 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2006,7 +2006,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); /* FLL pre-scaler */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, - NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + NAU8825_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8825_FLL_REF_DIV_SFT); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index b6b21b312854..514fd13c2f46 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -137,7 +137,8 @@ #define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ -#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) +#define NAU8825_FLL_REF_DIV_SFT 10 +#define NAU8825_FLL_REF_DIV_MASK (0x3 << NAU8825_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ #define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) From 4ee437fbf626b5ad756889d8bc0fcead3d66dde7 Mon Sep 17 00:00:00 2001 From: Caleb Crome Date: Tue, 3 Jan 2017 10:22:57 -0800 Subject: [PATCH 12/17] ASoC: fsl_ssi: set fifo watermark to more reliable value The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e. activate DMA on FIFO when only 2 spaces are left.) This means the DMA must service the fifo within 2 audio samples, which is just not enough time for many use cases with high data rate. In many configurations the audio channel slips (causing l/r swap in stereo configurations, or channel slipping in multi-channel configurations). This patch gives more breathing room and allows the SSI to operate reliably by changing the fifio refill watermark to 8. There is no change in behavior for older chips (with an 8-deep fifo). Only the newer chips with a 15-deep fifo get the new behavior. I suspect a new fifo depth setting could be optimized on the older chips too, but I have not tested. Signed-off-by: Caleb Crome Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++------------ 1 file changed, 53 insertions(+), 21 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 50349437d961..fde08660b63b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -224,6 +224,12 @@ struct fsl_ssi_soc_data { * @dbg_stats: Debugging statistics * * @soc: SoC specific data + * + * @fifo_watermark: the FIFO watermark setting. Notifies DMA when + * there are @fifo_watermark or fewer words in TX fifo or + * @fifo_watermark or more empty words in RX fifo. + * @dma_maxburst: max number of words to transfer in one go. So far, + * this is always the same as fifo_watermark. */ struct fsl_ssi_private { struct regmap *regs; @@ -263,6 +269,9 @@ struct fsl_ssi_private { const struct fsl_ssi_soc_data *soc; struct device *dev; + + u32 fifo_watermark; + u32 dma_maxburst; }; /* @@ -1051,21 +1060,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, regmap_write(regs, CCSR_SSI_SRCR, srcr); regmap_write(regs, CCSR_SSI_SCR, scr); - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't - * use FIFO 1. We program the transmit water to signal a DMA transfer - * if there are only two (or fewer) elements left in the FIFO. Two - * elements equals one frame (left channel, right channel). This value, - * however, depends on the depth of the transmit buffer. - * - * We set the watermark on the same level as the DMA burstsize. For - * fiq it is probably better to use the biggest possible watermark - * size. - */ - if (ssi_private->use_dma) - wm = ssi_private->fifo_depth - 2; - else - wm = ssi_private->fifo_depth; + wm = ssi_private->fifo_watermark; regmap_write(regs, CCSR_SSI_SFCSR, CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) | @@ -1373,12 +1368,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); - /* - * We have burstsize be "fifo_depth - 2" to match the SSI - * watermark setting in fsl_ssi_startup(). - */ - ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.maxburst = ssi_private->dma_maxburst; + ssi_private->dma_params_rx.maxburst = ssi_private->dma_maxburst; ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; @@ -1543,6 +1534,47 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + /* + * Set the watermark for transmit FIFO 0 and receive FIFO 0. We don't + * use FIFO 1 but set the watermark appropriately nontheless. + * We program the transmit water to signal a DMA transfer + * if there are N elements left in the FIFO. For chips with 15-deep + * FIFOs, set watermark to 8. This allows the SSI to operate at a + * high data rate without channel slipping. Behavior is unchanged + * for the older chips with a fifo depth of only 8. A value of 4 + * might be appropriate for the older chips, but is left at + * fifo_depth-2 until sombody has a chance to test. + * + * We set the watermark on the same level as the DMA burstsize. For + * fiq it is probably better to use the biggest possible watermark + * size. + */ + switch (ssi_private->fifo_depth) { + case 15: + /* + * 2 samples is not enough when running at high data + * rates (like 48kHz @ 16 bits/channel, 16 channels) + * 8 seems to split things evenly and leave enough time + * for the DMA to fill the FIFO before it's over/under + * run. + */ + ssi_private->fifo_watermark = 8; + ssi_private->dma_maxburst = 8; + break; + case 8: + default: + /* + * maintain old behavior for older chips. + * Keeping it the same because I don't have an older + * board to test with. + * I suspect this could be changed to be something to + * leave some more space in the fifo. + */ + ssi_private->fifo_watermark = ssi_private->fifo_depth - 2; + ssi_private->dma_maxburst = ssi_private->fifo_depth - 2; + break; + } + dev_set_drvdata(&pdev->dev, ssi_private); if (ssi_private->soc->imx) { From 60448b077ed93d227e6c117a9e87db76ff0c1911 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 4 Jan 2017 15:44:52 -0600 Subject: [PATCH 13/17] ASoC: Intel: bytcr-rt5640: fix settings in internal clock mode Frequency value of zero did not make sense, use same 24.576MHz setting and only change the clock source in idle mode Suggested-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e33e4777a65c..8d2fb2d6f532 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -142,7 +142,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, * for Jack detection and button press */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_RCCLK, - 0, + 48000 * 512, SND_SOC_CLOCK_IN); if (!ret) { if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && priv->mclk) From bc65a326c579e93a5c2120a65ede72f11369ee5a Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Mon, 2 Jan 2017 09:50:05 +0530 Subject: [PATCH 14/17] ASoC: Intel: Skylake: Release FW ctx in cleanup Saved firmware ctx was not never released, so release Firmware ctx in cleanup routine. Signed-off-by: Jeeja KP Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-sst.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 8fc3178bc79c..b30bd384c8d3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -515,6 +515,9 @@ EXPORT_SYMBOL_GPL(skl_sst_init_fw); void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) { + + if (ctx->dsp->fw) + release_firmware(ctx->dsp->fw); skl_clear_module_table(ctx->dsp); skl_freeup_uuid_list(ctx); skl_ipc_free(&ctx->ipc); From 9f169b9f52a4afccdab7a7d2311b0c53a78a1e6b Mon Sep 17 00:00:00 2001 From: Patrick Lai Date: Sat, 31 Dec 2016 22:44:39 -0800 Subject: [PATCH 15/17] ASoC: dpcm: Avoid putting stream state to STOP when FE stream is paused When multiple front-ends are using the same back-end, putting state of a front-end to STOP state upon receiving pause command will result in backend stream getting released by DPCM framework unintentionally. In order to avoid backend to be released when another active front-end stream is present, put the stream state to PAUSED state instead of STOP state. Signed-off-by: Patrick Lai Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7a1eaa2772f..6aba14009c92 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2184,9 +2184,11 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; } out: From 02c5c03283c52157d336abf5e44ffcda10579fbf Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Dec 2016 12:05:05 +0800 Subject: [PATCH 16/17] ASoC: rt5645: set sel_i2s_pre_div1 to 2 The i2s clock pre-divider 1 is used for both i2s1 and sysclk. The i2s1 is usually used for the main i2s and the pre-divider will be set in hw_params function. However, if i2s2 is used, the pre-divider is not set in the hw_params function and the default value of i2s clock pre-divider 1 is too high for sysclk and DMIC usage. Fix by overriding default divider value to 2. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681 Tested-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 10c2a564a715..1ac96ef9ee20 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3833,6 +3833,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, + RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); + if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); From 2e40795c3bf344cfb5220d94566205796e3ef19a Mon Sep 17 00:00:00 2001 From: Dennis Kadioglu Date: Mon, 9 Jan 2017 17:10:46 +0100 Subject: [PATCH 17/17] ALSA: usb-audio: Add a quirk for Plantronics BT600 Plantronics BT600 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x1" and "cannot get freq at ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b3fd2382fdd9..eb4b9f7a571e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1135,6 +1135,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0x02F7): /* Plantronics BT-600 */ case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */