From bd0b609e0c3362cb167c51d4bd4330d79fc00987 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Nov 2019 14:48:33 +0100 Subject: ASoC: core: Fix compile warning with CONFIG_DEBUG_FS=n MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Paper over a compile warning: sound/soc/soc-pcm.c:1185:8: warning: unused variable ‘name’ Fixes: 0632fa042541 ("ASoC: core: Fix pcm code debugfs error") Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20191107134833.1502-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 493a2e80e893..4bf71e3211d8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1182,7 +1182,9 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; +#ifdef CONFIG_DEBUG_FS char *name; +#endif /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { -- cgit From 8cd73ce7ba6381feb9df763240dee3a6d4aff1ca Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 1 Nov 2019 12:09:14 -0500 Subject: ASoC: SOF: topology: fix missing NULL pointer check Add check to avoid possible NULL pointer dereference issue. This issue was reported by static analysis tools, we didn't face this issue but we can't rule it out either as a false positive. Reported-by: Keqiao Zhang Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191101170916.26517-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 143b8259a70a..6096731e89ce 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -150,6 +150,11 @@ static int sof_keyword_dapm_event(struct snd_soc_dapm_widget *w, /* get runtime PCM params using widget's stream name */ spcm = snd_sof_find_spcm_name(sdev, swidget->widget->sname); + if (!spcm) { + dev_err(sdev->dev, "error: cannot find PCM for %s\n", + swidget->widget->name); + return -EINVAL; + } /* process events */ switch (event) { -- cgit From 16299326a0cbbc88d4d6491a8ceebbfca81064c5 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 1 Nov 2019 12:09:15 -0500 Subject: ASoC: SOF: Intel: hda: set L1SEN on S0ix suspend Set L1SEN to make sure the system can enter S0ix, and restore it on resume. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191101170916.26517-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 15 +++++++++++++++ sound/soc/sof/intel/hda.h | 3 +++ 2 files changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 8cd5ecc01b62..4a4d318f97ff 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -478,9 +478,16 @@ static int hda_resume(struct snd_sof_dev *sdev, bool runtime_resume) int hda_dsp_resume(struct snd_sof_dev *sdev) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct pci_dev *pci = to_pci_dev(sdev->dev); if (sdev->s0_suspend) { + /* restore L1SEN bit */ + if (hda->l1_support_changed) + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, 0); + /* restore and disable the system wakeup */ pci_restore_state(pci); disable_irq_wake(pci->irq); @@ -518,11 +525,19 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) int hda_dsp_suspend(struct snd_sof_dev *sdev) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct hdac_bus *bus = sof_to_bus(sdev); struct pci_dev *pci = to_pci_dev(sdev->dev); int ret; if (sdev->s0_suspend) { + /* enable L1SEN to make sure the system can enter S0Ix */ + hda->l1_support_changed = + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, + HDA_VS_INTEL_EM2_L1SEN); + /* enable the system waking up via IPC IRQ */ enable_irq_wake(pci->irq); pci_save_state(pci); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 5ad73a34b09c..18d7e72bf9b7 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -408,6 +408,9 @@ struct sof_intel_hda_dev { int irq; + /* PM related */ + bool l1_support_changed;/* during suspend, is L1SEN changed or not */ + /* DMIC device */ struct platform_device *dmic_dev; }; -- cgit From fe965096c9495ddcf78ec163348105e2baf8d185 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 11 Nov 2019 15:50:48 +0800 Subject: ASoC: fsl_audmix: Add spin lock to protect tdms Audmix support two substream, When two substream start to run, the trigger function may be called by two substream in same time, that the priv->tdms may be updated wrongly. The expected priv->tdms is 0x3, but sometimes the result is 0x2, or 0x1. Fixes: be1df61cf06e ("ASoC: fsl: Add Audio Mixer CPU DAI driver") Signed-off-by: Shengjiu Wang Acked-by: Nicolin Chen Reviewed-by: Daniel Baluta Link: https://lore.kernel.org/r/1e706afe53fdd1fbbbc79277c48a98f8416ba873.1573458378.git.shengjiu.wang@nxp.com Signed-off-by: Mark Brown Cc: --- sound/soc/fsl/fsl_audmix.c | 6 ++++++ sound/soc/fsl/fsl_audmix.h | 1 + 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index c7e4e9757dce..a1db1bce330f 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -286,6 +286,7 @@ static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai); + unsigned long lock_flags; /* Capture stream shall not be handled */ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -295,12 +296,16 @@ static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&priv->lock, lock_flags); priv->tdms |= BIT(dai->driver->id); + spin_unlock_irqrestore(&priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&priv->lock, lock_flags); priv->tdms &= ~BIT(dai->driver->id); + spin_unlock_irqrestore(&priv->lock, lock_flags); break; default: return -EINVAL; @@ -491,6 +496,7 @@ static int fsl_audmix_probe(struct platform_device *pdev) return PTR_ERR(priv->ipg_clk); } + spin_lock_init(&priv->lock); platform_set_drvdata(pdev, priv); pm_runtime_enable(dev); diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h index 7812ffec45c5..479f05695d53 100644 --- a/sound/soc/fsl/fsl_audmix.h +++ b/sound/soc/fsl/fsl_audmix.h @@ -96,6 +96,7 @@ struct fsl_audmix { struct platform_device *pdev; struct regmap *regmap; struct clk *ipg_clk; + spinlock_t lock; /* Protect tdms */ u8 tdms; }; -- cgit From 461c623270e4fa9cbab44e3ac2f0ddab2842c034 Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:26 -0800 Subject: ASoC: rt5677: Load firmware via SPI using delayed work The firmware rt5677_elf_vad is an ELF binary obtained from request_firmware(). Sections of the ELF are loaded to the DSP via SPI. A model (e.g. en_us.mmap) can optionally be loaded to the DSP at RT5677_MODEL_ADDR to overwrite the baked-in model in rt5677_elf_vad. Then we switch to DSP mode, load firmware, and let DSP run. When a hotword is detected, an interrupt is fired and rt5677_irq() is called. When 'DSP VAD Switch' is turned off, the codec is set back to normal mode. The kcontrol 'DSP VAD Switch' is automatically enabled/disabled when the hotwording PCM stream is opened/closed. Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-2-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.c | 13 ++ sound/soc/codecs/rt5677.c | 301 +++++++++++++++++++++++++++++++++--------- sound/soc/codecs/rt5677.h | 6 +- 3 files changed, 253 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 3aa3ea7c2768..25e28be3722e 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -26,6 +26,7 @@ #include +#include "rt5677.h" #include "rt5677-spi.h" #define DRV_NAME "rt5677spi" @@ -111,10 +112,16 @@ static int rt5677_spi_pcm_close( struct snd_soc_component *component, struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *codec_component = + snd_soc_rtdcom_lookup(rtd, "rt5677"); + struct rt5677_priv *rt5677 = + snd_soc_component_get_drvdata(codec_component); struct rt5677_dsp *rt5677_dsp = snd_soc_component_get_drvdata(component); cancel_delayed_work_sync(&rt5677_dsp->copy_work); + rt5677->set_dsp_vad(codec_component, false); return 0; } @@ -154,9 +161,15 @@ static int rt5677_spi_prepare( struct snd_soc_component *component, struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *rt5677_component = + snd_soc_rtdcom_lookup(rtd, "rt5677"); + struct rt5677_priv *rt5677 = + snd_soc_component_get_drvdata(rt5677_component); struct rt5677_dsp *rt5677_dsp = snd_soc_component_get_drvdata(component); + rt5677->set_dsp_vad(rt5677_component, true); rt5677_dsp->dma_offset = 0; rt5677_dsp->avail_bytes = 0; return 0; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 315a3d39bc09..13f21bf4e04e 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -38,6 +38,10 @@ #define RT5677_DEVICE_ID 0x6327 +/* Register controlling boot vector */ +#define RT5677_DSP_BOOT_VECTOR 0x1801f090 +#define RT5677_MODEL_ADDR 0x5FFC9800 + #define RT5677_PR_RANGE_BASE (0xff + 1) #define RT5677_PR_SPACING 0x100 @@ -686,10 +690,8 @@ static int rt5677_dsp_mode_i2c_read( return ret; } -static void rt5677_set_dsp_mode(struct snd_soc_component *component, bool on) +static void rt5677_set_dsp_mode(struct rt5677_priv *rt5677, bool on) { - struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); - if (on) { regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, RT5677_PWR_DSP, RT5677_PWR_DSP); @@ -701,86 +703,234 @@ static void rt5677_set_dsp_mode(struct snd_soc_component *component, bool on) } } +static unsigned int rt5677_set_vad_source(struct rt5677_priv *rt5677) +{ + /* DMIC1 power = enabled + * DMIC CLK = 256 * fs / 12 + */ + regmap_update_bits(rt5677->regmap, RT5677_DMIC_CTRL1, + RT5677_DMIC_CLK_MASK, 5 << RT5677_DMIC_CLK_SFT); + + /* I2S pre divide 2 = /6 (clk_sys2) */ + regmap_update_bits(rt5677->regmap, RT5677_CLK_TREE_CTRL1, + RT5677_I2S_PD2_MASK, RT5677_I2S_PD2_6); + + /* DSP Clock = MCLK1 (bypassed PLL2) */ + regmap_write(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_BYPASS); + + /* SAD Threshold1 */ + regmap_write(rt5677->regmap, RT5677_VAD_CTRL2, 0x013f); + /* SAD Threshold2 */ + regmap_write(rt5677->regmap, RT5677_VAD_CTRL3, 0x0ae5); + /* SAD Sample Rate Converter = Up 6 (8K to 48K) + * SAD Output Sample Rate = Same as I2S + * SAD Threshold3 + */ + regmap_update_bits(rt5677->regmap, RT5677_VAD_CTRL4, + RT5677_VAD_OUT_SRC_RATE_MASK | RT5677_VAD_OUT_SRC_MASK | + RT5677_VAD_LV_DIFF_MASK, 0x7f << RT5677_VAD_LV_DIFF_SFT); + /* Minimum frame level within a pre-determined duration = 32 frames + * Bypass ADPCM Encoder/Decoder = Bypass ADPCM + * Automatic Push Data to SAD Buffer Once SAD Flag is triggered = enable + * SAD Buffer Over-Writing = enable + * SAD Buffer Pop Mode Control = disable + * SAD Buffer Push Mode Control = enable + * SAD Detector Control = enable + * SAD Function Control = enable + * SAD Function Reset = normal + */ + regmap_write(rt5677->regmap, RT5677_VAD_CTRL1, + RT5677_VAD_FUNC_RESET | RT5677_VAD_FUNC_ENABLE | + RT5677_VAD_DET_ENABLE | RT5677_VAD_BUF_PUSH | + RT5677_VAD_BUF_OW | RT5677_VAD_FG2ENC | + RT5677_VAD_ADPCM_BYPASS | 1 << RT5677_VAD_MIN_DUR_SFT); + + /* IRQ Source of VAD Jack Detection = enable */ + regmap_write(rt5677->regmap, RT5677_IRQ_CTRL2, 0x4000); + + /* Private register, no doc */ + regmap_update_bits(rt5677->regmap, RT5677_PR_BASE + RT5677_BIAS_CUR4, + 0x0f00, 0x0100); + + /* LDO2 output = 1.2V + * LDO1 output = 1.2V (LDO_IN = 1.8V) + */ + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK, + 5 << RT5677_LDO1_SEL_SFT | 5 << RT5677_LDO2_SEL_SFT); + + /* Codec core power = power on + * LDO1 power = power on + */ + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_CORE | RT5677_PWR_LDO1, + RT5677_PWR_CORE | RT5677_PWR_LDO1); + + /* Isolation for DCVDD4 = normal (set during probe) + * Isolation for DCVDD2 = normal (set during probe) + * Isolation for DSP = normal + * Isolation for Band 0~7 = disable + * Isolation for InBound 4~10 and OutBound 4~10 = disable + */ + regmap_write(rt5677->regmap, RT5677_PWR_DSP2, + RT5677_PWR_CORE_ISO | RT5677_PWR_DSP_ISO | + RT5677_PWR_SR7_ISO | RT5677_PWR_SR6_ISO | + RT5677_PWR_SR5_ISO | RT5677_PWR_SR4_ISO | + RT5677_PWR_SR3_ISO | RT5677_PWR_SR2_ISO | + RT5677_PWR_SR1_ISO | RT5677_PWR_SR0_ISO | + RT5677_PWR_MLT_ISO); + + /* System Band 0~7 = power on + * InBound 4~10 and OutBound 4~10 = power on + * DSP = power on + * DSP CPU = stop (will be set to "run" after firmware loaded) + */ + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, + RT5677_PWR_SR7 | RT5677_PWR_SR6 | + RT5677_PWR_SR5 | RT5677_PWR_SR4 | + RT5677_PWR_SR3 | RT5677_PWR_SR2 | + RT5677_PWR_SR1 | RT5677_PWR_SR0 | + RT5677_PWR_MLT | RT5677_PWR_DSP | + RT5677_PWR_DSP_CPU); + + return 0; +} + +static int rt5677_parse_and_load_dsp(struct rt5677_priv *rt5677, const u8 *buf, + unsigned int len) +{ + struct snd_soc_component *component = rt5677->component; + Elf32_Ehdr *elf_hdr; + Elf32_Phdr *pr_hdr; + Elf32_Half i; + int ret = 0; + + if (!buf || (len < sizeof(Elf32_Ehdr))) + return -ENOMEM; + + elf_hdr = (Elf32_Ehdr *)buf; +#ifndef EM_XTENSA +#define EM_XTENSA 94 +#endif + if (strncmp(elf_hdr->e_ident, ELFMAG, sizeof(ELFMAG) - 1)) + dev_err(component->dev, "Wrong ELF header prefix\n"); + if (elf_hdr->e_ehsize != sizeof(Elf32_Ehdr)) + dev_err(component->dev, "Wrong Elf header size\n"); + if (elf_hdr->e_machine != EM_XTENSA) + dev_err(component->dev, "Wrong DSP code file\n"); + + if (len < elf_hdr->e_phoff) + return -ENOMEM; + pr_hdr = (Elf32_Phdr *)(buf + elf_hdr->e_phoff); + for (i = 0; i < elf_hdr->e_phnum; i++) { + /* TODO: handle p_memsz != p_filesz */ + if (pr_hdr->p_paddr && pr_hdr->p_filesz) { + dev_info(component->dev, "Load 0x%x bytes to 0x%x\n", + pr_hdr->p_filesz, pr_hdr->p_paddr); + + ret = rt5677_spi_write(pr_hdr->p_paddr, + buf + pr_hdr->p_offset, + pr_hdr->p_filesz); + if (ret) + dev_err(component->dev, "Load firmware failed %d\n", + ret); + } + pr_hdr++; + } + return ret; +} + +static int rt5677_load_dsp_from_file(struct rt5677_priv *rt5677) +{ + const struct firmware *fwp; + struct device *dev = rt5677->component->dev; + int ret = 0; + + /* Load dsp firmware from rt5677_elf_vad file */ + ret = request_firmware(&fwp, "rt5677_elf_vad", dev); + if (ret) { + dev_err(dev, "Request rt5677_elf_vad failed %d\n", ret); + return ret; + } + dev_info(dev, "Requested rt5677_elf_vad (%zu)\n", fwp->size); + + ret = rt5677_parse_and_load_dsp(rt5677, fwp->data, fwp->size); + release_firmware(fwp); + return ret; +} + static int rt5677_set_dsp_vad(struct snd_soc_component *component, bool on) { struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); - static bool activity; - int ret; + rt5677->dsp_vad_en = on; if (!IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI)) return -ENXIO; - if (on && !activity) { - activity = true; + schedule_delayed_work(&rt5677->dsp_work, 0); + return 0; +} - regcache_cache_only(rt5677->regmap, false); - regcache_cache_bypass(rt5677->regmap, true); +static void rt5677_dsp_work(struct work_struct *work) +{ + struct rt5677_priv *rt5677 = + container_of(work, struct rt5677_priv, dsp_work.work); + static bool activity; + bool enable = rt5677->dsp_vad_en; - regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x1); - regmap_update_bits(rt5677->regmap, - RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0f00); - regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, - RT5677_LDO1_SEL_MASK, 0x0); - regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, - RT5677_PWR_LDO1, RT5677_PWR_LDO1); - switch (rt5677->type) { - case RT5677: - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | - RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | - RT5677_DSP_CLK_SRC_BYPASS); - break; - case RT5676: - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_DSP_CLK_SRC_MASK, - RT5677_DSP_CLK_SRC_BYPASS); - break; - default: - break; - } - regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); - regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); - rt5677_set_dsp_mode(component, true); - - ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, - component->dev); - if (ret == 0) { - rt5677_spi_write_firmware(0x50000000, rt5677->fw1); - release_firmware(rt5677->fw1); - } - ret = request_firmware(&rt5677->fw2, RT5677_FIRMWARE2, - component->dev); - if (ret == 0) { - rt5677_spi_write_firmware(0x60000000, rt5677->fw2); - release_firmware(rt5677->fw2); - } + dev_info(rt5677->component->dev, "DSP VAD: enable=%d, activity=%d\n", + enable, activity); - regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x0); + if (enable && !activity) { + activity = true; - regcache_cache_bypass(rt5677->regmap, false); - regcache_cache_only(rt5677->regmap, true); - } else if (!on && activity) { + /* Set GPIO1 as an output pin driving a 0. Firmware will + * raise GPIO1 upon hotword detect. + */ + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, + RT5677_GPIO1_DIR_MASK | RT5677_GPIO1_OUT_MASK | + RT5677_GPIO1_P_MASK, RT5677_GPIO1_DIR_OUT | + RT5677_GPIO1_OUT_LO | RT5677_GPIO1_P_NOR); + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, + RT5677_GPIO1_PIN_MASK, RT5677_GPIO1_PIN_GPIO1); + + rt5677_set_vad_source(rt5677); + rt5677_set_dsp_mode(rt5677, true); + + /* Boot the firmware from IRAM instead of SRAM0. */ + rt5677_dsp_mode_i2c_write_addr(rt5677, RT5677_DSP_BOOT_VECTOR, + 0x0009, 0x0003); + rt5677_dsp_mode_i2c_write_addr(rt5677, RT5677_DSP_BOOT_VECTOR, + 0x0019, 0x0003); + rt5677_dsp_mode_i2c_write_addr(rt5677, RT5677_DSP_BOOT_VECTOR, + 0x0009, 0x0003); + + rt5677_load_dsp_from_file(rt5677); + + /* Set DSP CPU to Run */ + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, + RT5677_PWR_DSP_CPU, 0x0); + } else if (!enable && activity) { activity = false; - regcache_cache_only(rt5677->regmap, false); - regcache_cache_bypass(rt5677->regmap, true); + /* Set DSP CPU to Stop */ + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, + RT5677_PWR_DSP_CPU, RT5677_PWR_DSP_CPU); - regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x1); - rt5677_set_dsp_mode(component, false); - regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x0001); + rt5677_set_dsp_mode(rt5677, false); - regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + /* Disable and clear VAD interrupt */ + regmap_write(rt5677->regmap, RT5677_VAD_CTRL1, 0x2184); + regmap_update_bits(rt5677->regmap, RT5677_IRQ_CTRL2, + 0xF000, 0x0000); - regcache_cache_bypass(rt5677->regmap, false); - regcache_mark_dirty(rt5677->regmap); - regcache_sync(rt5677->regmap); - } + /* Set GPIO1 pin back to be IRQ output for jack detect */ + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, + RT5677_GPIO1_PIN_MASK, RT5677_GPIO1_PIN_IRQ); - return 0; + } } static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); @@ -819,7 +969,8 @@ static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) - rt5677_set_dsp_vad(component, rt5677->dsp_vad_en); + rt5677_set_dsp_vad(component, + !!ucontrol->value.integer.value[0]); return 0; } @@ -4740,6 +4891,8 @@ static void rt5677_remove(struct snd_soc_component *component) { struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + cancel_delayed_work_sync(&rt5677->dsp_work); + regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); gpiod_set_value_cansleep(rt5677->reset_pin, 1); @@ -4938,6 +5091,17 @@ static struct snd_soc_dai_driver rt5677_dai[] = { }, .ops = &rt5677_aif_dai_ops, }, + { + .name = "rt5677-dspbuffer", + .id = RT5677_DSPBUFF, + .capture = { + .stream_name = "DSP Buffer", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, }; static const struct snd_soc_component_driver soc_component_dev_rt5677 = { @@ -5081,6 +5245,9 @@ static irqreturn_t rt5677_irq(int unused, void *data) mutex_lock(&rt5677->irq_lock); + if (rt5677->dsp_vad_en) + rt5677_spi_hotword_detected(); + /* * Loop to handle interrupts until the last i2c read shows no pending * irqs. The interrupt line is shared by multiple interrupt sources. @@ -5271,6 +5438,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c) return -ENOMEM; rt5677->dev = &i2c->dev; + rt5677->set_dsp_vad = rt5677_set_dsp_vad; + INIT_DELAYED_WORK(&rt5677->dsp_work, rt5677_dsp_work); i2c_set_clientdata(i2c, rt5677); if (i2c->dev.of_node) { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 213f4b8ca269..d18b41da1176 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1730,6 +1730,7 @@ enum { RT5677_AIF4, RT5677_AIF5, RT5677_AIFS, + RT5677_DSPBUFF, }; enum { @@ -1845,14 +1846,17 @@ struct rt5677_priv { #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif - bool dsp_vad_en; + bool dsp_vad_en; /* DSP VAD enable/disable request */ bool is_dsp_mode; bool is_vref_slow; + struct delayed_work dsp_work; /* Interrupt handling */ struct irq_domain *domain; struct mutex irq_lock; unsigned int irq_en; + + int (*set_dsp_vad)(struct snd_soc_component *component, bool on); }; int rt5677_sel_asrc_clk_src(struct snd_soc_component *component, -- cgit From 157b006f6be46be833b2941856b86e0a9d8bfb0e Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:27 -0800 Subject: ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device This link is needed for the RT5677 DSP to do hotwording Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-3-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d3353e26d..eeb9a11e4e14 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -74,6 +74,7 @@ static const struct snd_soc_dapm_route bdw_rt5677_map[] = { /* CODEC BE connections */ {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, + {"DSP Capture", NULL, "DSP Buffer"}, }; static const struct snd_kcontrol_new bdw_rt5677_controls[] = { @@ -258,6 +259,12 @@ SND_SOC_DAILINK_DEF(platform, SND_SOC_DAILINK_DEF(be, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-RT5677CE:00", "rt5677-aif1"))); +/* Wake on voice interface */ +SND_SOC_DAILINK_DEFS(dsp, + DAILINK_COMP_ARRAY(COMP_CPU("spi-RT5677AA:00")), + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-RT5677CE:00", "rt5677-dspbuffer")), + DAILINK_COMP_ARRAY(COMP_PLATFORM("spi-RT5677AA:00"))); + static struct snd_soc_dai_link bdw_rt5677_dais[] = { /* Front End DAI links */ { @@ -276,6 +283,13 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { SND_SOC_DAILINK_REG(fe, dummy, platform), }, + /* Non-DPCM links */ + { + .name = "Codec DSP", + .stream_name = "Wake on Voice", + SND_SOC_DAILINK_REG(dsp), + }, + /* Back End DAI links */ { /* SSP0 - Codec */ -- cgit From 21c00e5df4397870ee835c974bf50570f9d24253 Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:28 -0800 Subject: ASoC: rt5677: Enable jack detect while DSP is running Before a hotword is detected, GPIO1 pin is configured as IRQ output so that jack detect works. When a hotword is detected, the DSP firmware configures the GPIO1 pin as GPIO1 and drives a 1. rt5677_irq() is called after a rising edge on the GPIO1 pin, due to either jack detect event or hotword event, or both. All possible events are checked and handled in rt5677_irq() where GPIO1 pin is configured back to IRQ output if a hotword is detected. Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-4-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 64 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 48 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 13f21bf4e04e..48955b22262f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -312,6 +312,8 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_IRQ_CTRL1: case RT5677_IRQ_CTRL2: case RT5677_GPIO_ST: + case RT5677_GPIO_CTRL1: /* Modified by DSP firmware */ + case RT5677_GPIO_CTRL2: /* Modified by DSP firmware */ case RT5677_DSP_INB1_SRC_CTRL4: case RT5677_DSP_INB2_SRC_CTRL4: case RT5677_DSP_INB3_SRC_CTRL4: @@ -746,8 +748,11 @@ static unsigned int rt5677_set_vad_source(struct rt5677_priv *rt5677) RT5677_VAD_BUF_OW | RT5677_VAD_FG2ENC | RT5677_VAD_ADPCM_BYPASS | 1 << RT5677_VAD_MIN_DUR_SFT); - /* IRQ Source of VAD Jack Detection = enable */ - regmap_write(rt5677->regmap, RT5677_IRQ_CTRL2, 0x4000); + /* VAD/SAD is not routed to the IRQ output (i.e. MX-BE[14] = 0), but it + * is routed to DSP_IRQ_0, so DSP firmware may use it to sleep and save + * power. See ALC5677 datasheet section 9.17 "GPIO, Interrupt and Jack + * Detection" for more info. + */ /* Private register, no doc */ regmap_update_bits(rt5677->regmap, RT5677_PR_BASE + RT5677_BIAS_CUR4, @@ -886,15 +891,15 @@ static void rt5677_dsp_work(struct work_struct *work) if (enable && !activity) { activity = true; - /* Set GPIO1 as an output pin driving a 0. Firmware will - * raise GPIO1 upon hotword detect. + /* Before a hotword is detected, GPIO1 pin is configured as IRQ + * output so that jack detect works. When a hotword is detected, + * the DSP firmware configures the GPIO1 pin as GPIO1 and + * drives a 1. rt5677_irq() is called after a rising edge on + * the GPIO1 pin, due to either jack detect event or hotword + * event, or both. All possible events are checked and handled + * in rt5677_irq() where GPIO1 pin is configured back to IRQ + * output if a hotword is detected. */ - regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2, - RT5677_GPIO1_DIR_MASK | RT5677_GPIO1_OUT_MASK | - RT5677_GPIO1_P_MASK, RT5677_GPIO1_DIR_OUT | - RT5677_GPIO1_OUT_LO | RT5677_GPIO1_P_NOR); - regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, - RT5677_GPIO1_PIN_MASK, RT5677_GPIO1_PIN_GPIO1); rt5677_set_vad_source(rt5677); rt5677_set_dsp_mode(rt5677, true); @@ -915,6 +920,8 @@ static void rt5677_dsp_work(struct work_struct *work) } else if (!enable && activity) { activity = false; + /* Don't turn off the DSP while handling irqs */ + mutex_lock(&rt5677->irq_lock); /* Set DSP CPU to Stop */ regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, RT5677_PWR_DSP_CPU, RT5677_PWR_DSP_CPU); @@ -923,13 +930,12 @@ static void rt5677_dsp_work(struct work_struct *work) /* Disable and clear VAD interrupt */ regmap_write(rt5677->regmap, RT5677_VAD_CTRL1, 0x2184); - regmap_update_bits(rt5677->regmap, RT5677_IRQ_CTRL2, - 0xF000, 0x0000); /* Set GPIO1 pin back to be IRQ output for jack detect */ regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, RT5677_GPIO1_PIN_MASK, RT5677_GPIO1_PIN_IRQ); + mutex_unlock(&rt5677->irq_lock); } } @@ -5237,6 +5243,28 @@ static const struct rt5677_irq_desc rt5677_irq_descs[] = { }, }; +bool rt5677_check_hotword(struct rt5677_priv *rt5677) +{ + int reg_gpio; + + if (!rt5677->is_dsp_mode) + return false; + + if (regmap_read(rt5677->regmap, RT5677_GPIO_CTRL1, ®_gpio)) + return false; + + /* Firmware sets GPIO1 pin to be GPIO1 after hotword is detected */ + if ((reg_gpio & RT5677_GPIO1_PIN_MASK) == RT5677_GPIO1_PIN_IRQ) + return false; + + /* Set GPIO1 pin back to be IRQ output for jack detect */ + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, + RT5677_GPIO1_PIN_MASK, RT5677_GPIO1_PIN_IRQ); + + rt5677_spi_hotword_detected(); + return true; +} + static irqreturn_t rt5677_irq(int unused, void *data) { struct rt5677_priv *rt5677 = data; @@ -5245,9 +5273,6 @@ static irqreturn_t rt5677_irq(int unused, void *data) mutex_lock(&rt5677->irq_lock); - if (rt5677->dsp_vad_en) - rt5677_spi_hotword_detected(); - /* * Loop to handle interrupts until the last i2c read shows no pending * irqs. The interrupt line is shared by multiple interrupt sources. @@ -5285,7 +5310,13 @@ static irqreturn_t rt5677_irq(int unused, void *data) reg_irq ^= rt5677_irq_descs[i].polarity_mask; } } - if (!irq_fired) + + /* Exit the loop only when we know for sure that GPIO1 pin + * was low at some point since irq_lock was acquired. Any event + * after that point creates a rising edge that triggers another + * call to rt5677_irq(). + */ + if (!irq_fired && !rt5677_check_hotword(rt5677)) goto exit; ret = regmap_write(rt5677->regmap, RT5677_IRQ_CTRL1, reg_irq); @@ -5296,6 +5327,7 @@ static irqreturn_t rt5677_irq(int unused, void *data) } } exit: + WARN_ON_ONCE(loop == 20); mutex_unlock(&rt5677->irq_lock); if (irq_fired) return IRQ_HANDLED; -- cgit From 29073ae40c472f17d42aa38850da861b5e3f912e Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:29 -0800 Subject: ASoC: rt5677: Add DAPM audio path for hotword stream Add a DAPM audio path from "DMIC L1" to "DSP Buffer" so that when hotwording is enabled, DAPM does not power off the codec with SND_SOC_BIAS_OFF. Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-5-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 48955b22262f..ea235f3874ca 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -707,6 +707,13 @@ static void rt5677_set_dsp_mode(struct rt5677_priv *rt5677, bool on) static unsigned int rt5677_set_vad_source(struct rt5677_priv *rt5677) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(rt5677->component); + /* Force dapm to sync before we enable the + * DSP to prevent write corruption + */ + snd_soc_dapm_sync(dapm); + /* DMIC1 power = enabled * DMIC CLK = 256 * fs / 12 */ @@ -3167,6 +3174,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SLBRX", "SLIMBus Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SLBTX", "SLIMBus Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DSPTX", "DSP Buffer", 0, SND_SOC_NOPM, 0, 0), /* Sidetone Mux */ SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, @@ -3701,11 +3709,24 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "SLBTX", NULL, "SLB ADC3 Mux" }, { "SLBTX", NULL, "SLB ADC4 Mux" }, + { "DSPTX", NULL, "IB01 Bypass Mux" }, + { "IB01 Mux", "IF1 DAC 01", "IF1 DAC01" }, { "IB01 Mux", "IF2 DAC 01", "IF2 DAC01" }, { "IB01 Mux", "SLB DAC 01", "SLB DAC01" }, { "IB01 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, - { "IB01 Mux", "VAD ADC/DAC1 FS", "DAC1 FS" }, + /* The IB01 Mux controls the source for InBound0 and InBound1. + * When the mux option "VAD ADC/DAC1 FS" is selected, "VAD ADC" goes to + * InBound0 and "DAC1 FS" goes to InBound1. "VAD ADC" is used for + * hotwording. "DAC1 FS" is not used currently. + * + * Creating a common widget node for "VAD ADC" + "DAC1 FS" and + * connecting the common widget to IB01 Mux causes the issue where + * there is an active path going from system playback -> "DAC1 FS" -> + * IB01 Mux -> DSP Buffer -> hotword stream. This wrong path confuses + * DAPM. Therefore "DAC1 FS" is ignored for now. + */ + { "IB01 Mux", "VAD ADC/DAC1 FS", "VAD ADC Mux" }, { "IB01 Bypass Mux", "Bypass", "IB01 Mux" }, { "IB01 Bypass Mux", "Pass SRC", "IB01 Mux" }, -- cgit From eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:30 -0800 Subject: ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1 while it's streaming audio over SPI. The DSP firmware turns on PLL2 (MX-64 bit 8) when SPI streaming starts. However regmap does not believe that register can change by itself. When BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't read the register first before write, so PLL2 power bit is cleared by accident. Marking MX-64h as volatile in regmap solved the issue. Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ea235f3874ca..e5db9dc60378 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -302,6 +302,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_I2C_MASTER_CTRL7: case RT5677_I2C_MASTER_CTRL8: case RT5677_HAP_GENE_CTRL2: + case RT5677_PWR_ANLG2: /* Modified by DSP firmware */ case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: case RT5677_ASRC_22: -- cgit From 3f81068dc8e5fc0d3b0d67a5ae99c5140a4e96cb Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:31 -0800 Subject: ASoC: rt5677: Stop and restart DSP over suspend/resume MCLK1 gets disabled at suspend and re-enabled at resume. Before MCLK1 is re-enabled, if the DSP is already on (either the DSP was left on during suspend, or the DSP is turned on early at resume), i2c register read returns garbage and corrupts the regmap cache. This patch stops the DSP before suspend and restarts it after resume with a dalay to ensure MCLK is on while loading firmware. Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-7-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 24 +++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 3 ++- 2 files changed, 23 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index e5db9dc60378..eb55d6b9d0c1 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -876,6 +876,7 @@ static int rt5677_load_dsp_from_file(struct rt5677_priv *rt5677) static int rt5677_set_dsp_vad(struct snd_soc_component *component, bool on) { struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + rt5677->dsp_vad_en_request = on; rt5677->dsp_vad_en = on; if (!IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI)) @@ -969,7 +970,7 @@ static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); - ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; + ucontrol->value.integer.value[0] = rt5677->dsp_vad_en_request; return 0; } @@ -4636,14 +4637,15 @@ static int rt5677_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + enum snd_soc_bias_level prev_bias = + snd_soc_component_get_bias_level(component); switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_STANDBY) { - rt5677_set_dsp_vad(component, false); + if (prev_bias == SND_SOC_BIAS_STANDBY) { regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK, @@ -4667,9 +4669,25 @@ static int rt5677_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: + if (prev_bias == SND_SOC_BIAS_OFF && + rt5677->dsp_vad_en_request) { + /* Re-enable the DSP if it was turned off at suspend */ + rt5677->dsp_vad_en = true; + /* The delay is to wait for MCLK */ + schedule_delayed_work(&rt5677->dsp_work, + msecs_to_jiffies(1000)); + } break; case SND_SOC_BIAS_OFF: + flush_delayed_work(&rt5677->dsp_work); + if (rt5677->is_dsp_mode) { + /* Turn off the DSP before suspend */ + rt5677->dsp_vad_en = false; + schedule_delayed_work(&rt5677->dsp_work, 0); + flush_delayed_work(&rt5677->dsp_work); + } + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0); regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000); regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d18b41da1176..046ed2ee8e31 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1846,7 +1846,8 @@ struct rt5677_priv { #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif - bool dsp_vad_en; /* DSP VAD enable/disable request */ + bool dsp_vad_en_request; /* DSP VAD enable/disable request */ + bool dsp_vad_en; /* dsp_work parameter */ bool is_dsp_mode; bool is_vref_slow; struct delayed_work dsp_work; -- cgit From ee0be4a99d4923a9b9c681b008df561648731793 Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Tue, 5 Nov 2019 17:13:32 -0800 Subject: ASoC: rt5677: Disable irq at suspend The irq is disabled at suspend to avoid running the threaded irq handler after the codec has been powered off. At resume, codec irq is re-enabled and the interrupt status register is checked to see if headphone has been pluggnd/unplugged while the device is suspended. There is still a chance that the headphone gets enabled or disabled after the codec is suspended. disable_irq syncs the threaded irq handler, but soc-jack's threaded irq handler schedules a delayed work to poll gpios (for debounce). This is still OK. The codec won't be powered back on again because all audio paths have been suspended, and there are no force enabled supply widgets (MICBIAS1 is disabled). The gpio status read after codec power off could be wrong, so the gpio values are checked again after resume. Signed-off-by: Ben Zhang Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-8-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5677.h | 2 ++ 2 files changed, 48 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index eb55d6b9d0c1..14f04db8e5e7 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4949,6 +4949,11 @@ static int rt5677_suspend(struct snd_soc_component *component) { struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + if (rt5677->irq) { + cancel_delayed_work_sync(&rt5677->resume_irq_check); + disable_irq(rt5677->irq); + } + if (!rt5677->dsp_vad_en) { regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); @@ -4977,6 +4982,11 @@ static int rt5677_resume(struct snd_soc_component *component) regcache_sync(rt5677->regmap); } + if (rt5677->irq) { + enable_irq(rt5677->irq); + schedule_delayed_work(&rt5677->resume_irq_check, 0); + } + return 0; } #else @@ -5375,6 +5385,39 @@ exit: return IRQ_NONE; } +static void rt5677_resume_irq_check(struct work_struct *work) +{ + int i, virq; + struct rt5677_priv *rt5677 = + container_of(work, struct rt5677_priv, resume_irq_check.work); + + /* This is needed to check and clear the interrupt status register + * at resume. If the headset is plugged/unplugged when the device is + * fully suspended, there won't be a rising edge at resume to trigger + * the interrupt. Without this, we miss the next unplug/plug event. + */ + rt5677_irq(0, rt5677); + + /* Call all enabled jack detect irq handlers again. This is needed in + * addition to the above check for a corner case caused by jack gpio + * debounce. After codec irq is disabled at suspend, the delayed work + * scheduled by soc-jack may run and read wrong jack gpio values, since + * the regmap is in cache only mode. At resume, there is no irq because + * rt5677_irq has already ran and cleared the irq status at suspend. + * Without this explicit check, unplug the headset right after suspend + * starts, then after resume the headset is still shown as plugged in. + */ + mutex_lock(&rt5677->irq_lock); + for (i = 0; i < RT5677_IRQ_NUM; i++) { + if (rt5677->irq_en & rt5677_irq_descs[i].enable_mask) { + virq = irq_find_mapping(rt5677->domain, i); + if (virq) + handle_nested_irq(virq); + } + } + mutex_unlock(&rt5677->irq_lock); +} + static void rt5677_irq_bus_lock(struct irq_data *data) { struct rt5677_priv *rt5677 = irq_data_get_irq_chip_data(data); @@ -5450,6 +5493,7 @@ static int rt5677_init_irq(struct i2c_client *i2c) } mutex_init(&rt5677->irq_lock); + INIT_DELAYED_WORK(&rt5677->resume_irq_check, rt5677_resume_irq_check); /* * Select RC as the debounce clock so that GPIO works even when @@ -5495,6 +5539,8 @@ static int rt5677_init_irq(struct i2c_client *i2c) if (ret) dev_err(&i2c->dev, "Failed to request IRQ: %d\n", ret); + rt5677->irq = i2c->irq; + return ret; } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 046ed2ee8e31..f8ada967fdbc 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1856,6 +1856,8 @@ struct rt5677_priv { struct irq_domain *domain; struct mutex irq_lock; unsigned int irq_en; + struct delayed_work resume_irq_check; + int irq; int (*set_dsp_vad)(struct snd_soc_component *component, bool on); }; -- cgit From 395f02ef210e6f7dad2ca0efc16ffaea4a9ed030 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 5 Nov 2019 17:13:33 -0800 Subject: ASoC: rt5677: Allow VAD to be shut on/off at all times Due to limitations of the clocking configuration, we have no way of scheduling our hibernation before the bdw dsp hibernates. This causes issues when the system suspends with an open stream. We need userspace to toggle the kcontrol before we are suspended so that any writes on suspend are not lost and we don't corrupt the regmap. Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-9-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 14f04db8e5e7..a65d1554366d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -979,13 +979,8 @@ static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); - - rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; - if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) - rt5677_set_dsp_vad(component, - !!ucontrol->value.integer.value[0]); + rt5677_set_dsp_vad(component, !!ucontrol->value.integer.value[0]); return 0; } -- cgit From 55229597a94531726878229ccfcd3fe4ec572dc3 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 5 Nov 2019 17:13:34 -0800 Subject: ASoC: bdw-rt5677: Turn on MCLK1 for DSP via DAPM The RT5677 DSP needs the I2S MCLK1 to run its DSP. Add a dapm route to SSP0 CODEC IN so the clock is turned on automatically when the DSP is turned on. Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-10-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw-rt5677.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index eeb9a11e4e14..b2475e3eff7b 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -75,6 +75,10 @@ static const struct snd_soc_dapm_route bdw_rt5677_map[] = { {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, {"DSP Capture", NULL, "DSP Buffer"}, + + /* DSP Clock Connections */ + { "DSP Buffer", NULL, "SSP0 CODEC IN" }, + { "SSP0 CODEC IN", NULL, "DSPTX" }, }; static const struct snd_kcontrol_new bdw_rt5677_controls[] = { -- cgit From ba0b3a977ecf525231d36f2d9f3a6ea05c35090a Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 5 Nov 2019 17:13:35 -0800 Subject: ASoC: rt5677: Set ADC clock to use PLL and enable ASRC Use the PLL to kept the correct 24M clock rate so frequency shift does not occur when using the DSP VAD. Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-11-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 6 ++++++ sound/soc/codecs/rt5677.h | 2 ++ sound/soc/intel/boards/bdw-rt5677.c | 33 +++++++++++++++++++++++++++++++++ 3 files changed, 41 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index a65d1554366d..0e7773584145 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5046,6 +5046,11 @@ static const struct snd_soc_dai_ops rt5677_aif_dai_ops = { .set_tdm_slot = rt5677_set_tdm_slot, }; +static const struct snd_soc_dai_ops rt5677_dsp_dai_ops = { + .set_sysclk = rt5677_set_dai_sysclk, + .set_pll = rt5677_set_dai_pll, +}; + static struct snd_soc_dai_driver rt5677_dai[] = { { .name = "rt5677-aif1", @@ -5152,6 +5157,7 @@ static struct snd_soc_dai_driver rt5677_dai[] = { .rates = SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .ops = &rt5677_dsp_dai_ops, }, }; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index f8ada967fdbc..944ae02aafc2 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1336,6 +1336,8 @@ #define RT5677_PLL_M_SFT 12 #define RT5677_PLL_M_BP (0x1 << 11) #define RT5677_PLL_M_BP_SFT 11 +#define RT5677_PLL_UPDATE_PLL1 (0x1 << 1) +#define RT5677_PLL_UPDATE_PLL1_SFT 1 /* Global Clock Control 1 (0x80) */ #define RT5677_SCLK_SRC_MASK (0x3 << 14) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index b2475e3eff7b..2af8e5a62da8 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -170,10 +170,37 @@ static int bdw_rt5677_hw_params(struct snd_pcm_substream *substream, return ret; } +static int bdw_rt5677_dsp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_PLL1, 24576000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5677_PLL1_S_MCLK, + 24000000, 24576000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll configuration\n"); + return ret; + } + + return 0; +} + static const struct snd_soc_ops bdw_rt5677_ops = { .hw_params = bdw_rt5677_hw_params, }; +static const struct snd_soc_ops bdw_rt5677_dsp_ops = { + .hw_params = bdw_rt5677_dsp_hw_params, +}; + #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) static int bdw_rt5677_rtd_init(struct snd_soc_pcm_runtime *rtd) { @@ -213,6 +240,11 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) rt5677_sel_asrc_clk_src(component, RT5677_DA_STEREO_FILTER | RT5677_AD_STEREO1_FILTER | RT5677_I2S1_SOURCE, RT5677_CLK_SEL_I2S1_ASRC); + /* Enable codec ASRC function for Mono ADC L. + * The ASRC clock source is clk_sys2_asrc. + */ + rt5677_sel_asrc_clk_src(component, RT5677_AD_MONO_L_FILTER, + RT5677_CLK_SEL_SYS2); /* Request rt5677 GPIO for headphone amp control */ bdw_rt5677->gpio_hp_en = devm_gpiod_get(component->dev, "headphone-enable", @@ -291,6 +323,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { { .name = "Codec DSP", .stream_name = "Wake on Voice", + .ops = &bdw_rt5677_dsp_ops, SND_SOC_DAILINK_REG(dsp), }, -- cgit From 9da776ba7852f9b5cbfdfaa80a1dc07cda592b55 Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Tue, 5 Nov 2019 17:13:36 -0800 Subject: ASoC: rt5677: Wait for DSP to boot before loading firmware Wait for hardware to startup. If we load before hardware is ready we could end up corrupting the firmware. Signed-off-by: Curtis Malainey Link: https://lore.kernel.org/r/20191106011335.223061-12-cujomalainey@chromium.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 0e7773584145..f2f763b4c399 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -892,6 +892,7 @@ static void rt5677_dsp_work(struct work_struct *work) container_of(work, struct rt5677_priv, dsp_work.work); static bool activity; bool enable = rt5677->dsp_vad_en; + int i, val; dev_info(rt5677->component->dev, "DSP VAD: enable=%d, activity=%d\n", @@ -913,6 +914,18 @@ static void rt5677_dsp_work(struct work_struct *work) rt5677_set_vad_source(rt5677); rt5677_set_dsp_mode(rt5677, true); +#define RT5677_BOOT_RETRY 20 + for (i = 0; i < RT5677_BOOT_RETRY; i++) { + regmap_read(rt5677->regmap, RT5677_PWR_DSP_ST, &val); + if (val == 0x3ff) + break; + udelay(500); + } + if (i == RT5677_BOOT_RETRY && val != 0x3ff) { + dev_err(rt5677->component->dev, "DSP Boot Timed Out!"); + return; + } + /* Boot the firmware from IRAM instead of SRAM0. */ rt5677_dsp_mode_i2c_write_addr(rt5677, RT5677_DSP_BOOT_VECTOR, 0x0009, 0x0003); -- cgit From 36da67630d3159e48e37a9f9b2d26ceece426482 Mon Sep 17 00:00:00 2001 From: Sebastian Reichel Date: Fri, 8 Nov 2019 18:48:40 +0100 Subject: ASoC: Add DA7213 audio codec as selectable option This commit adds the Dialog DA7213 audio codec as a selectable option in the kernel config. Currently the driver can only be selected for Intel Baytrail/Cherrytrail devices or if SND_SOC_ALL_CODECS is enabled. Signed-off-by: Sebastian Reichel Link: https://lore.kernel.org/r/20191108174843.11227-3-sebastian.reichel@collabora.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1a4b084cc90d..ec01e4f12a78 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -679,7 +679,8 @@ config SND_SOC_DA7210 tristate config SND_SOC_DA7213 - tristate + tristate "Dialog DA7213 CODEC" + depends on I2C config SND_SOC_DA7218 tristate -- cgit From e2db787bdcb4f2722ecf410168f0583764634e45 Mon Sep 17 00:00:00 2001 From: Yu-Hsuan Hsu Date: Tue, 24 Sep 2019 00:29:40 +0800 Subject: ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint On KBL platform, the microphone is attached to external codec(rt5514) instead of PCH. However, TDM slot between PCH and codec is 16 bits only. In order to avoid setting wrong format, we should add a constraint to force to use 16 bits format forever. Signed-off-by: Yu-Hsuan Hsu Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index a1056cda3dd7..3e5f6bead229 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -465,6 +465,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- cgit From 13409d27cb39fb1dfcb28418a9bc3d26907e1dbc Mon Sep 17 00:00:00 2001 From: Michael Walle Date: Fri, 8 Nov 2019 21:31:52 +0100 Subject: ASoC: wm8904: configure sysclk/FLL automatically This adds a new mode WM8904_CLK_AUTO which automatically enables the FLL if a frequency different than the MCLK is set. These additions make the codec work with the simple-card driver in general and especially in systems where the MCLK doesn't match the required clock. Signed-off-by: Michael Walle Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20191108203152.19098-1-michael@walle.cc Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 72 +++++++++++++++++++++++++++++------------------ sound/soc/codecs/wm8904.h | 1 + 2 files changed, 45 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index bcb3c9d5abf0..2a7d23a5daa8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1410,34 +1410,6 @@ static int wm8904_hw_params(struct snd_pcm_substream *substream, return 0; } - -static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, - unsigned int freq, int dir) -{ - struct snd_soc_component *component = dai->component; - struct wm8904_priv *priv = snd_soc_component_get_drvdata(component); - - switch (clk_id) { - case WM8904_CLK_MCLK: - priv->sysclk_src = clk_id; - priv->mclk_rate = freq; - break; - - case WM8904_CLK_FLL: - priv->sysclk_src = clk_id; - break; - - default: - return -EINVAL; - } - - dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); - - wm8904_configure_clocking(component); - - return 0; -} - static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; @@ -1824,6 +1796,50 @@ out: return 0; } +static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct wm8904_priv *priv = snd_soc_component_get_drvdata(component); + unsigned long mclk_freq; + int ret; + + switch (clk_id) { + case WM8904_CLK_AUTO: + mclk_freq = clk_get_rate(priv->mclk); + /* enable FLL if a different sysclk is desired */ + if (mclk_freq != freq) { + priv->sysclk_src = WM8904_CLK_FLL; + ret = wm8904_set_fll(dai, WM8904_FLL_MCLK, + WM8904_FLL_MCLK, + mclk_freq, freq); + if (ret) + return ret; + break; + } + clk_id = WM8904_CLK_MCLK; + /* fallthrough */ + + case WM8904_CLK_MCLK: + priv->sysclk_src = clk_id; + priv->mclk_rate = freq; + break; + + case WM8904_CLK_FLL: + priv->sysclk_src = clk_id; + break; + + default: + return -EINVAL; + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + wm8904_configure_clocking(component); + + return 0; +} + static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_component *component = codec_dai->component; diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h index c1bca52f9927..de6340446b1f 100644 --- a/sound/soc/codecs/wm8904.h +++ b/sound/soc/codecs/wm8904.h @@ -10,6 +10,7 @@ #ifndef _WM8904_H #define _WM8904_H +#define WM8904_CLK_AUTO 0 #define WM8904_CLK_MCLK 1 #define WM8904_CLK_FLL 2 -- cgit From e7cfd867fd9842f346688f28412eb83dec342900 Mon Sep 17 00:00:00 2001 From: Jacob Rasmussen Date: Mon, 11 Nov 2019 11:59:57 -0700 Subject: ASoC: rt5645: Fixed buddy jack support. The headphone jack on buddy was broken with the following commit: commit 6b5da66322c5 ("ASoC: rt5645: read jd1_1 status for jd detection"). This changes the jd_mode for buddy to 4 so buddy can read from the same register that was used in the working version of this driver without affecting any other devices that might use this, since no other device uses jd_mode = 4. To test this I plugged and uplugged the headphone jack, verifying audio works. Signed-off-by: Jacob Rasmussen Reviewed-by: Ross Zwisler Link: https://lore.kernel.org/r/20191111185957.217244-1-jacobraz@google.com Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1c06b3b9218c..902ac98a3fbe 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3270,6 +3270,9 @@ static void rt5645_jack_detect_work(struct work_struct *work) snd_soc_jack_report(rt5645->mic_jack, report, SND_JACK_MICROPHONE); return; + case 4: + val = snd_soc_component_read32(rt5645->component, RT5645_A_JD_CTRL1) & 0x002; + break; default: /* read rt5645 jd1_1 status */ val = snd_soc_component_read32(rt5645->component, RT5645_INT_IRQ_ST) & 0x1000; break; @@ -3603,7 +3606,7 @@ static const struct rt5645_platform_data intel_braswell_platform_data = { static const struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, - .jd_mode = 3, + .jd_mode = 4, .level_trigger_irq = true, }; @@ -3999,6 +4002,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, RT5645_JD1_MODE_1); break; case 3: + case 4: regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, RT5645_JD1_MODE_MASK, RT5645_JD1_MODE_2); -- cgit From 24de63562b9da9fb3145329abb226ab623e6af19 Mon Sep 17 00:00:00 2001 From: Bard liao Date: Mon, 11 Nov 2019 16:21:52 -0600 Subject: ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null jack_detect_work will be triggered by rt5682_irq. We should cancel it if hs_jack is set to null. Signed-off-by: Bard liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111222152.19723-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c50b75ce82e0..62b8ed412bd1 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1002,6 +1002,7 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, RT5682_JD1_EN_MASK, RT5682_JD1_DIS); regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); return 0; } -- cgit From 130d3e90777fe974e4b8fa100cec8faf19cac998 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 11 Nov 2019 16:29:00 -0600 Subject: ASoC: SOF: Intel: Fix CFL and CML FW nocodec binary names. The manifest information is different between CNL, CML and CFL platforms hence we need to load different files. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111222901.19892-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..2ef927371b23 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -113,7 +113,7 @@ static const struct sof_dev_desc cnl_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) static const struct sof_dev_desc cfl_desc = { - .machines = snd_soc_acpi_intel_cnl_machines, + .machines = snd_soc_acpi_intel_cfl_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -122,7 +122,7 @@ static const struct sof_dev_desc cfl_desc = { .chip_info = &cnl_chip_info, .default_fw_path = "intel/sof", .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-cnl.ri", + .nocodec_fw_filename = "sof-cfl.ri", .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .arch_ops = &sof_xtensa_arch_ops @@ -133,7 +133,7 @@ static const struct sof_dev_desc cfl_desc = { IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) static const struct sof_dev_desc cml_desc = { - .machines = snd_soc_acpi_intel_cnl_machines, + .machines = snd_soc_acpi_intel_cml_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -142,7 +142,7 @@ static const struct sof_dev_desc cml_desc = { .chip_info = &cnl_chip_info, .default_fw_path = "intel/sof", .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-cnl.ri", + .nocodec_fw_filename = "sof-cml.ri", .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .arch_ops = &sof_xtensa_arch_ops -- cgit From 1f24d93c4f4ec0bb0b73c5b020060aa2d9faa756 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 11 Nov 2019 16:28:59 -0600 Subject: ASoC: Intel: acpi-match: split CNL tables in three Due to firmware manifest/signature differences, we have to use different firmware names, so split CNL machine table in three (CNL, CFL, CML). The CFL table is currently empty since all known platforms use HDaudio, but let's plan ahead. Reviewed-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111222901.19892-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-cfl-match.c | 18 ++++++++ sound/soc/intel/common/soc-acpi-intel-cml-match.c | 56 +++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 38 --------------- 4 files changed, 76 insertions(+), 39 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-cfl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cml-match.c (limited to 'sound') diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index eafe95ead49b..bd352878f89a 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-hsw-bdw-match.o \ soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ - soc-acpi-intel-cnl-match.o soc-acpi-intel-icl-match.o \ + soc-acpi-intel-cnl-match.o soc-acpi-intel-cfl-match.o \ + soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o \ soc-acpi-intel-hda-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-cfl-match.c b/sound/soc/intel/common/soc-acpi-intel-cfl-match.c new file mode 100644 index 000000000000..d6fd2026d0b8 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-cfl-match.c @@ -0,0 +1,18 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-apci-intel-cfl-match.c - tables and support for CFL ACPI enumeration. + * + * Copyright (c) 2019, Intel Corporation. + * + */ + +#include +#include + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cfl_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c new file mode 100644 index 000000000000..5d08ae066738 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -0,0 +1,56 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * soc-acpi-intel-cml-match.c - tables and support for CML ACPI enumeration. + * + * Copyright (c) 2019, Intel Corporation. + * + */ + +#include +#include + +static struct snd_soc_acpi_codecs cml_codecs = { + .num_codecs = 1, + .codecs = {"10EC5682"} +}; + +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { + { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { + .id = "MX98357A", + .drv_name = "sof_rt5682", + .quirk_data = &cml_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", + }, + { + .id = "10EC1011", + .drv_name = "cml_rt1011_rt5682", + .quirk_data = &cml_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", + }, + { + .id = "10EC5682", + .drv_name = "sof_rt5682", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-rt5682.tplg", + }, + + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index 16d0bae8b316..27588841c8b0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -14,16 +14,6 @@ static struct skl_machine_pdata cnl_pdata = { .use_tplg_pcm = true, }; -static struct snd_soc_acpi_codecs cml_codecs = { - .num_codecs = 1, - .codecs = {"10EC5682"} -}; - -static struct snd_soc_acpi_codecs cml_spk_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -33,34 +23,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_fw_filename = "sof-cnl.ri", .sof_tplg_filename = "sof-cnl-rt274.tplg", }, - { - .id = "DLGS7219", - .drv_name = "cml_da7219_max98357a", - .quirk_data = &cml_spk_codecs, - .sof_fw_filename = "sof-cnl.ri", - .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", - }, - { - .id = "MX98357A", - .drv_name = "sof_rt5682", - .quirk_data = &cml_codecs, - .sof_fw_filename = "sof-cnl.ri", - .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", - }, - { - .id = "10EC1011", - .drv_name = "cml_rt1011_rt5682", - .quirk_data = &cml_codecs, - .sof_fw_filename = "sof-cnl.ri", - .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", - }, - { - .id = "10EC5682", - .drv_name = "sof_rt5682", - .sof_fw_filename = "sof-cnl.ri", - .sof_tplg_filename = "sof-cml-rt5682.tplg", - }, - {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_machines); -- cgit From 0af237d51a4e734db959a158185f79019f022f59 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 11 Nov 2019 16:29:01 -0600 Subject: ASoC: SOF: Intel: hda: use fallback for firmware name We have platforms such as CFL with no known I2S codec being used, and the ACPI tables are currently empty, so fall-back to using the firmware filename used in nocodec mode Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111222901.19892-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7dc0018dc4c3..91bd88fddac7 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -415,9 +415,16 @@ static int hda_init_caps(struct snd_sof_dev *sdev) pdata->tplg_filename = hda_mach->sof_tplg_filename; - /* firmware: pick the first in machine list */ + /* + * firmware: pick the first in machine list, + * or use nocodec firmware name if list is empty + */ mach = pdata->desc->machines; - pdata->fw_filename = mach->sof_fw_filename; + if (mach->id[0]) + pdata->fw_filename = mach->sof_fw_filename; + else + pdata->fw_filename = + pdata->desc->nocodec_fw_filename; dev_info(bus->dev, "using HDA machine driver %s now\n", hda_mach->drv_name); -- cgit From 74b4dd04b19cc03e1cc369d9c371cf4206367df4 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Mon, 11 Nov 2019 16:33:40 -0600 Subject: ASoC: SOF: PM: add state machine to comments Add Audio DSP state machine with comments. Note that the 'D0<-->runtime D0I3' part is not implemented yet. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111223343.19986-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 560a937e0484..3204c92f0e7d 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -428,6 +428,38 @@ int snd_sof_set_d0_substate(struct snd_sof_dev *sdev, } EXPORT_SYMBOL(snd_sof_set_d0_substate); +/* + * Audio DSP states may transform as below:- + * + * D0I3 compatible stream + * Runtime +---------------------+ opened only, timeout + * suspend | +--------------------+ + * +------------+ D0(active) | | + * | | <---------------+ | + * | +--------> | | | + * | |Runtime +--^--+---------^--+--+ The last | | + * | |resume | | | | opened D0I3 | | + * | | | | | | compatible | | + * | | resume| | | | stream closed | | + * | | from | | D3 | | | | + * | | D3 | |suspend | | d0i3 | | + * | | | | | |suspend | | + * | | | | | | | | + * | | | | | | | | + * +-v---+-----------+--v-------+ | | +------+----v----+ + * | | | +-----------> | + * | D3 (suspended) | | | D0I3 +-----+ + * | | +--------------+ | | + * | | resume from | | | + * +-------------------^--------+ d0i3 suspend +----------------+ | + * | | + * | D3 suspend | + * +------------------------------------------------+ + * + * d0i3_suspend = s0_suspend && D0I3 stream opened, + * D3 suspend = !d0i3_suspend, + */ + int snd_sof_resume(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); -- cgit From 58a972efd21bb5adadd97f2546693ec64bfc99b9 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Mon, 11 Nov 2019 16:33:41 -0600 Subject: ASoC: SOF: PM: add check before setting d0_substate Add check before seeting d0_substate and return success if Audio DSP is already in the target substate. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111223343.19986-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 3204c92f0e7d..b89810bf3230 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -416,6 +416,9 @@ int snd_sof_set_d0_substate(struct snd_sof_dev *sdev, { int ret; + if (sdev->d0_substate == d0_substate) + return 0; + /* do platform specific set_state */ ret = snd_sof_dsp_set_power_state(sdev, d0_substate); if (ret < 0) -- cgit From a77e5d573f847d2f984c38eb711351326bc55258 Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Mon, 11 Nov 2019 16:33:42 -0600 Subject: ASoC: SOF: add helper to check if we should enter d0i3 suspend Add helper to check if the DSP should be put in D0i3. This function returns true if a stream has ignored the SUSPEND trigger to keep the pipelines running in the DSP. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111223343.19986-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 13 +++++++++++++ sound/soc/sof/sof-priv.h | 2 ++ 2 files changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 8661c2cca76b..805918d3bcc0 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -132,6 +132,19 @@ struct snd_sof_dai *snd_sof_find_dai(struct snd_sof_dev *sdev, return NULL; } +bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev) +{ + struct snd_sof_pcm *spcm; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].suspend_ignored || + spcm->stream[SNDRV_PCM_STREAM_CAPTURE].suspend_ignored) + return true; + } + + return false; +} + /* * FW Panic/fault handling. */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 6408ac88a3e5..c7c2c70ee4d0 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -553,6 +553,8 @@ struct snd_sof_pcm *snd_sof_find_spcm_dai(struct snd_sof_dev *sdev, return NULL; } +bool snd_sof_dsp_d0i3_on_suspend(struct snd_sof_dev *sdev); + struct snd_sof_pcm *snd_sof_find_spcm_name(struct snd_sof_dev *sdev, const char *name); struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_sof_dev *sdev, -- cgit From 5e35d5f4224adb24c2bfa814b9fb562050aba3ea Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Mon, 11 Nov 2019 16:33:43 -0600 Subject: ASoC: SOF: PM: only suspend to D0I3 when needed We should suspend audio to D3 by default, for the sake of power saving, change the condition of D0I3 suspending here to that when there is stream with suspend_ignored specified. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111223343.19986-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index b89810bf3230..0fd5567237a8 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -468,7 +468,7 @@ int snd_sof_resume(struct device *dev) struct snd_sof_dev *sdev = dev_get_drvdata(dev); int ret; - if (sdev->s0_suspend) { + if (snd_sof_dsp_d0i3_on_suspend(sdev)) { /* resume from D0I3 */ dev_dbg(sdev->dev, "DSP will exit from D0i3...\n"); ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I0); @@ -497,7 +497,7 @@ int snd_sof_suspend(struct device *dev) struct snd_sof_dev *sdev = dev_get_drvdata(dev); int ret; - if (sdev->s0_suspend) { + if (snd_sof_dsp_d0i3_on_suspend(sdev)) { /* suspend to D0i3 */ dev_dbg(sdev->dev, "DSP is trying to enter D0i3...\n"); ret = snd_sof_set_d0_substate(sdev, SOF_DSP_D0I3); -- cgit From 1b4efdaf6d6053c8944cee0edba0969dc1be7d4b Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Mon, 11 Nov 2019 16:20:38 -0600 Subject: ASoC: SOF: topology: free kcontrol memory on error The volume and bytes kcontrols are currently not freeing their memory on initialization failures. When an error occurs, all the widgets loaded so far are unloaded via sof_widget_unload(). But this only happens for the widgets that got successfully loaded. Fix that by kfree()-ing the allocated memory on load error. Fixes: 311ce4fe7637d ("ASoC: SOF: Add support for loading topologies") Reviewed-by: Paul Olaru Reviewed-by: Ranjani Sridharan Signed-off-by: Dragos Tarcatu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111222039.19651-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 67 +++++++++++++++++++++++++++++++++++------------- 1 file changed, 49 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 6096731e89ce..d82ab981e840 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -942,18 +942,22 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, struct sof_ipc_ctrl_data *cdata; int tlv[TLV_ITEMS]; unsigned int i; - int ret; + int ret = 0; /* validate topology data */ - if (le32_to_cpu(mc->num_channels) > SND_SOC_TPLG_MAX_CHAN) - return -EINVAL; + if (le32_to_cpu(mc->num_channels) > SND_SOC_TPLG_MAX_CHAN) { + ret = -EINVAL; + goto out; + } /* init the volume get/put data */ scontrol->size = struct_size(scontrol->control_data, chanv, le32_to_cpu(mc->num_channels)); scontrol->control_data = kzalloc(scontrol->size, GFP_KERNEL); - if (!scontrol->control_data) - return -ENOMEM; + if (!scontrol->control_data) { + ret = -ENOMEM; + goto out; + } scontrol->comp_id = sdev->next_comp_id; scontrol->min_volume_step = le32_to_cpu(mc->min); @@ -963,7 +967,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, /* set cmd for mixer control */ if (le32_to_cpu(mc->max) == 1) { scontrol->cmd = SOF_CTRL_CMD_SWITCH; - goto out; + goto skip; } scontrol->cmd = SOF_CTRL_CMD_VOLUME; @@ -971,14 +975,15 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, /* extract tlv data */ if (get_tlv_data(kc->tlv.p, tlv) < 0) { dev_err(sdev->dev, "error: invalid TLV data\n"); - return -EINVAL; + ret = -EINVAL; + goto out_free; } /* set up volume table */ ret = set_up_volume_table(scontrol, tlv, le32_to_cpu(mc->max) + 1); if (ret < 0) { dev_err(sdev->dev, "error: setting up volume table\n"); - return ret; + goto out_free; } /* set default volume values to 0dB in control */ @@ -988,7 +993,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, cdata->chanv[i].value = VOL_ZERO_DB; } -out: +skip: /* set up possible led control from mixer private data */ ret = sof_parse_tokens(scomp, &scontrol->led_ctl, led_tokens, ARRAY_SIZE(led_tokens), mc->priv.array, @@ -996,13 +1001,21 @@ out: if (ret != 0) { dev_err(sdev->dev, "error: parse led tokens failed %d\n", le32_to_cpu(mc->priv.size)); - return ret; + goto out_free_table; } dev_dbg(sdev->dev, "tplg: load kcontrol index %d chans %d\n", scontrol->comp_id, scontrol->num_channels); - return 0; + return ret; + +out_free_table: + if (le32_to_cpu(mc->max) > 1) + kfree(scontrol->volume_table); +out_free: + kfree(scontrol->control_data); +out: + return ret; } static int sof_control_load_enum(struct snd_soc_component *scomp, @@ -1047,6 +1060,7 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, container_of(hdr, struct snd_soc_tplg_bytes_control, hdr); struct soc_bytes_ext *sbe = (struct soc_bytes_ext *)kc->private_value; int max_size = sbe->max; + int ret = 0; /* init the get/put bytes data */ scontrol->size = sizeof(struct sof_ipc_ctrl_data) + @@ -1055,13 +1069,16 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, if (scontrol->size > max_size) { dev_err(sdev->dev, "err: bytes data size %d exceeds max %d.\n", scontrol->size, max_size); - return -EINVAL; + ret = -EINVAL; + goto out; } scontrol->control_data = kzalloc(max_size, GFP_KERNEL); cdata = scontrol->control_data; - if (!scontrol->control_data) - return -ENOMEM; + if (!scontrol->control_data) { + ret = -ENOMEM; + goto out; + } scontrol->comp_id = sdev->next_comp_id; scontrol->cmd = SOF_CTRL_CMD_BINARY; @@ -1076,23 +1093,32 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, if (cdata->data->magic != SOF_ABI_MAGIC) { dev_err(sdev->dev, "error: Wrong ABI magic 0x%08x.\n", cdata->data->magic); - return -EINVAL; + ret = -EINVAL; + goto out_free; } if (SOF_ABI_VERSION_INCOMPATIBLE(SOF_ABI_VERSION, cdata->data->abi)) { dev_err(sdev->dev, "error: Incompatible ABI version 0x%08x.\n", cdata->data->abi); - return -EINVAL; + ret = -EINVAL; + goto out_free; } if (cdata->data->size + sizeof(const struct sof_abi_hdr) != le32_to_cpu(control->priv.size)) { dev_err(sdev->dev, "error: Conflict in bytes vs. priv size.\n"); - return -EINVAL; + ret = -EINVAL; + goto out_free; } } - return 0; + + return ret; + +out_free: + kfree(scontrol->control_data); +out: + return ret; } /* external kcontrol init - used for any driver specific init */ @@ -1150,6 +1176,11 @@ static int sof_control_load(struct snd_soc_component *scomp, int index, return 0; } + if (ret < 0) { + kfree(scontrol); + return ret; + } + dobj->private = scontrol; list_add(&scontrol->list, &sdev->kcontrol_list); return ret; -- cgit From 8f157d4ff039e03e2ed4cb602eeed2fd4687a58f Mon Sep 17 00:00:00 2001 From: Pawel Harlozinski Date: Tue, 12 Nov 2019 14:02:36 +0100 Subject: ASoC: Jack: Fix NULL pointer dereference in snd_soc_jack_report Check for existance of jack before tracing. NULL pointer dereference has been reported by KASAN while unloading machine driver (snd_soc_cnl_rt274). Signed-off-by: Pawel Harlozinski Link: https://lore.kernel.org/r/20191112130237.10141-1-pawel.harlozinski@linux.intel.com Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-jack.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index a71d2340eb05..b5748dcd490f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -82,10 +82,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - trace_snd_soc_jack_report(jack, mask, status); - if (!jack) return; + trace_snd_soc_jack_report(jack, mask, status); dapm = &jack->card->dapm; -- cgit From 0d18a7caa654ea1a0c02b3a253adfd5c10723871 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:07 +0900 Subject: ASoC: soc-core: remove soc_is_dai_link_bound() Because complex separeted "card pre-listed component" and "topology added component" duplicated operation is now becoming simple, we don't need to check already bound dai_link which is not exist anymore. This patch removes soc_is_dai_link_bound(). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87v9rxhkxw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 19 ------------------- 1 file changed, 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 55014e7ae0d8..c6885adbdc8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -953,19 +953,6 @@ struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_find_dai_link); -static bool soc_is_dai_link_bound(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - struct snd_soc_pcm_runtime *rtd; - - for_each_card_rtds(card, rtd) { - if (rtd->dai_link == dai_link) - return true; - } - - return false; -} - static int soc_dai_link_sanity_check(struct snd_soc_card *card, struct snd_soc_dai_link *link) { @@ -1085,12 +1072,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); - if (soc_is_dai_link_bound(card, dai_link)) { - dev_dbg(card->dev, "ASoC: dai link %s already bound\n", - dai_link->name); - return 0; - } - ret = soc_dai_link_sanity_check(card, dai_link); if (ret < 0) return ret; -- cgit From 7ca24386a7c2fb3828303b7c694cb0b4af1eac5c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:17 +0900 Subject: ASoC: soc-core: merge snd_soc_register_dai() and soc_add_dai() We don't need to separete snd_soc_register_dai() and soc_add_dai() anymore. Let's merge these Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87tv7hhkxm.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 43 ++++++++++++++++--------------------------- 1 file changed, 16 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c6885adbdc8b..b2544c7ff0f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2517,16 +2517,27 @@ static void soc_del_dai(struct snd_soc_dai *dai) list_del(&dai->list); } -/* Create a DAI and add it to the component's DAI list */ -static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, - bool legacy_dai_naming) +/** + * snd_soc_register_dai - Register a DAI dynamically & create its widgets + * + * @component: The component the DAIs are registered for + * @dai_drv: DAI driver to use for the DAI + * + * Topology can use this API to register DAIs when probing a component. + * These DAIs's widgets will be freed in the card cleanup and the DAIs + * will be freed in the component cleanup. + */ +struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_dai *dai; dev_dbg(dev, "ASoC: dynamically register DAI %s\n", dev_name(dev)); + lockdep_assert_held(&client_mutex); + dai = devm_kzalloc(dev, sizeof(*dai), GFP_KERNEL); if (dai == NULL) return NULL; @@ -2565,6 +2576,7 @@ static struct snd_soc_dai *soc_add_dai(struct snd_soc_component *component, dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); return dai; } +EXPORT_SYMBOL_GPL(snd_soc_register_dai); void snd_soc_unregister_dai(struct snd_soc_dai *dai) { @@ -2572,29 +2584,6 @@ void snd_soc_unregister_dai(struct snd_soc_dai *dai) } EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); -/** - * snd_soc_register_dai - Register a DAI dynamically & create its widgets - * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAI - * - * Topology can use this API to register DAIs when probing a component. - * These DAIs's widgets will be freed in the card cleanup and the DAIs - * will be freed in the component cleanup. - */ -struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, - bool legacy_dai_naming) -{ - struct device *dev = component->dev; - - dev_dbg(dev, "ASoC: dai register %s\n", dai_drv->name); - - lockdep_assert_held(&client_mutex); - return soc_add_dai(component, dai_drv, legacy_dai_naming); -} -EXPORT_SYMBOL_GPL(snd_soc_register_dai); - /** * snd_soc_unregister_dai - Unregister DAIs from the ASoC core * -- cgit From ffdbca0be6c78ea32b9243eea976270441210f2f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:23 +0900 Subject: ASoC: soc-core: merge snd_soc_unregister_dai() and soc_del_dai() We don't need to separete snd_soc_unregister_dai() and soc_del_dai() anymore. Let's merge these Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87sgn1hkxg.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b2544c7ff0f4..01a8fb28b48f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2511,11 +2511,12 @@ static inline char *fmt_multiple_name(struct device *dev, return devm_kstrdup(dev, dai_drv->name, GFP_KERNEL); } -static void soc_del_dai(struct snd_soc_dai *dai) +void snd_soc_unregister_dai(struct snd_soc_dai *dai) { dev_dbg(dai->dev, "ASoC: Unregistered DAI '%s'\n", dai->name); list_del(&dai->list); } +EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); /** * snd_soc_register_dai - Register a DAI dynamically & create its widgets @@ -2576,13 +2577,6 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); return dai; } -EXPORT_SYMBOL_GPL(snd_soc_register_dai); - -void snd_soc_unregister_dai(struct snd_soc_dai *dai) -{ - soc_del_dai(dai); -} -EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); /** * snd_soc_unregister_dai - Unregister DAIs from the ASoC core -- cgit From 63dc47da1f396fecd2373e41928e275f9ca3d924 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:31 +0900 Subject: ASoC: soc-core: merge snd_soc_add_dai_link() and soc_bind_dai_link() We don't need to separete snd_soc_add_dai_link() and soc_bind_dai_link() anymore. Let's merge these. One note is that before this patch, it adds list (A) eventhough if it had dai_link->ignore (1), or already bounded dai_link (2). But I guess it is wrong. This patch also solve this issue. /* BEFORE */ int soc_bind_dai_link(...) { ... (1) if (dai_link->ignore) return 0; (2) if (soc_is_dai_link_bound(...)) return 0; ... } int snd_soc_add_dai_link(...) { ... => ret = soc_bind_dai_link(...); => if (ret < 0) => return ret; (A) list_add_tail(&dai_link->list, &card->dai_link_list); ... } /* AFTER */ int snd_soc_add_dai_link(...) { ... (1) if (dai_link->ignore) return 0; (2) if (soc_is_dai_link_bound(...)) return 0; ... (A) list_add_tail(&dai_link->list, &card->dai_link_list); return 0; } Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87r22lhkx8.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 62 +++++++++++++++++++++------------------------------- 1 file changed, 25 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 01a8fb28b48f..8add98431881 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1059,14 +1059,33 @@ static void soc_unbind_dai_link(struct snd_soc_card *card, soc_free_pcm_runtime(rtd); } -static int soc_bind_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) +/** + * snd_soc_add_dai_link - Add a DAI link dynamically + * @card: The ASoC card to which the DAI link is added + * @dai_link: The new DAI link to add + * + * This function adds a DAI link to the ASoC card's link list. + * + * Note: Topology can use this API to add DAI links when probing the + * topology component. And machine drivers can still define static + * DAI links in dai_link array. + */ +int snd_soc_add_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codec, *platform; struct snd_soc_component *component; int i, ret; + lockdep_assert_held(&client_mutex); + + /* + * Notify the machine driver for extra initialization + */ + if (card->add_dai_link) + card->add_dai_link(card, dai_link); + if (dai_link->ignore) return 0; @@ -1115,12 +1134,16 @@ static int soc_bind_dai_link(struct snd_soc_card *card, } } + /* see for_each_card_links */ + list_add_tail(&dai_link->list, &card->dai_link_list); + return 0; _err_defer: soc_free_pcm_runtime(rtd); return -EPROBE_DEFER; } +EXPORT_SYMBOL_GPL(snd_soc_add_dai_link); static void soc_set_of_name_prefix(struct snd_soc_component *component) { @@ -1411,41 +1434,6 @@ void snd_soc_disconnect_sync(struct device *dev) } EXPORT_SYMBOL_GPL(snd_soc_disconnect_sync); -/** - * snd_soc_add_dai_link - Add a DAI link dynamically - * @card: The ASoC card to which the DAI link is added - * @dai_link: The new DAI link to add - * - * This function adds a DAI link to the ASoC card's link list. - * - * Note: Topology can use this API to add DAI links when probing the - * topology component. And machine drivers can still define static - * DAI links in dai_link array. - */ -int snd_soc_add_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - int ret; - - lockdep_assert_held(&client_mutex); - - /* - * Notify the machine driver for extra initialization - */ - if (card->add_dai_link) - card->add_dai_link(card, dai_link); - - ret = soc_bind_dai_link(card, dai_link); - if (ret < 0) - return ret; - - /* see for_each_card_links */ - list_add_tail(&dai_link->list, &card->dai_link_list); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_add_dai_link); - /** * snd_soc_remove_dai_link - Remove a DAI link from the list * @card: The ASoC card that owns the link -- cgit From da704f26ba376bd93ac5234fa4605c4a8e4a5648 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:38 +0900 Subject: ASoC: soc-core: merge snd_soc_remove_dai_link() and soc_unbind_dai_link() We don't need to separete snd_soc_remove_dai_link() and soc_unbind_dai_link() anymore. Let's merge these. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87pni5hkx1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 52 +++++++++++++++++++++++----------------------------- 1 file changed, 23 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8add98431881..d80d485f46d1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1049,15 +1049,36 @@ static int soc_dai_link_sanity_check(struct snd_soc_card *card, return 0; } -static void soc_unbind_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) +/** + * snd_soc_remove_dai_link - Remove a DAI link from the list + * @card: The ASoC card that owns the link + * @dai_link: The DAI link to remove + * + * This function removes a DAI link from the ASoC card's link list. + * + * For DAI links previously added by topology, topology should + * remove them by using the dobj embedded in the link. + */ +void snd_soc_remove_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; + lockdep_assert_held(&client_mutex); + + /* + * Notify the machine driver for extra destruction + */ + if (card->remove_dai_link) + card->remove_dai_link(card, dai_link); + + list_del(&dai_link->list); + rtd = snd_soc_get_pcm_runtime(card, dai_link->name); if (rtd) soc_free_pcm_runtime(rtd); } +EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); /** * snd_soc_add_dai_link - Add a DAI link dynamically @@ -1434,33 +1455,6 @@ void snd_soc_disconnect_sync(struct device *dev) } EXPORT_SYMBOL_GPL(snd_soc_disconnect_sync); -/** - * snd_soc_remove_dai_link - Remove a DAI link from the list - * @card: The ASoC card that owns the link - * @dai_link: The DAI link to remove - * - * This function removes a DAI link from the ASoC card's link list. - * - * For DAI links previously added by topology, topology should - * remove them by using the dobj embedded in the link. - */ -void snd_soc_remove_dai_link(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - lockdep_assert_held(&client_mutex); - - /* - * Notify the machine driver for extra destruction - */ - if (card->remove_dai_link) - card->remove_dai_link(card, dai_link); - - list_del(&dai_link->list); - - soc_unbind_dai_link(card, dai_link); -} -EXPORT_SYMBOL_GPL(snd_soc_remove_dai_link); - static int soc_link_dai_pcm_new(struct snd_soc_dai **dais, int num_dais, struct snd_soc_pcm_runtime *rtd) { -- cgit From c6619b729814c855fc7bfa5a6936f5ea94d60dfd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:46 +0900 Subject: ASoC: soc-core: merge soc_remove_component() and soc_cleanup_component() having both soc_remove_component() and soc_cleanup_component() is very confusable. Let's merge these. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87o8xphkwt.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d80d485f46d1..11cbd7915154 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1201,8 +1201,16 @@ static void soc_set_name_prefix(struct snd_soc_card *card, soc_set_of_name_prefix(component); } -static void soc_cleanup_component(struct snd_soc_component *component) +static void soc_remove_component(struct snd_soc_component *component, + int probed) { + + if (!component->card) + return; + + if (probed) + snd_soc_component_remove(component); + /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); @@ -1213,22 +1221,13 @@ static void soc_cleanup_component(struct snd_soc_component *component) snd_soc_component_module_put_when_remove(component); } -static void soc_remove_component(struct snd_soc_component *component) -{ - if (!component->card) - return; - - snd_soc_component_remove(component); - - soc_cleanup_component(component); -} - static int soc_probe_component(struct snd_soc_card *card, struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct snd_soc_dai *dai; + int probed = 0; int ret; if (!strcmp(component->name, "snd-soc-dummy")) @@ -1284,6 +1283,7 @@ static int soc_probe_component(struct snd_soc_card *card, dapm->bias_level != SND_SOC_BIAS_OFF, "codec %s can not start from non-off bias with idle_bias_off==1\n", component->name); + probed = 1; /* machine specific init */ if (component->init) { @@ -1312,7 +1312,7 @@ static int soc_probe_component(struct snd_soc_card *card, err_probe: if (ret < 0) - soc_cleanup_component(component); + soc_remove_component(component, probed); return ret; } @@ -1414,7 +1414,7 @@ static void soc_remove_link_components(struct snd_soc_card *card) if (component->driver->remove_order != order) continue; - soc_remove_component(component); + soc_remove_component(component, 1); } } } @@ -1608,7 +1608,7 @@ static void soc_remove_aux_devices(struct snd_soc_card *card) for_each_comp_order(order) { for_each_card_auxs_safe(card, comp, _comp) { if (comp->driver->remove_order == order) - soc_remove_component(comp); + soc_remove_component(comp, 1); } } } -- cgit From ed90c013a773b5f9e06089d0eed6714761152d14 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:07:56 +0900 Subject: ASoC: soc-core: merge snd_soc_bind_card() and snd_soc_instantiate_card() having both soc_bind_card() and snd_soc_instantiate_card() is very confusable. Let's merge these. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87mud9hkwj.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 42 ++++++++++++++---------------------------- 1 file changed, 14 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 11cbd7915154..f3d33a908fbc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1974,7 +1974,7 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) card->remove(card); } -static int snd_soc_instantiate_card(struct snd_soc_card *card) +static int snd_soc_bind_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link *dai_link; @@ -2106,6 +2106,19 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); + /* deactivate pins to sleep state */ + for_each_card_rtds(card, rtd) { + struct snd_soc_dai *dai; + + for_each_rtd_codec_dai(rtd, i, dai) { + if (!dai->active) + pinctrl_pm_select_sleep_state(dai->dev); + } + + if (!rtd->cpu_dai->active) + pinctrl_pm_select_sleep_state(rtd->cpu_dai->dev); + } + probe_end: if (ret < 0) soc_cleanup_card_resources(card); @@ -2338,33 +2351,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls); -static int snd_soc_bind_card(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - int ret; - - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - for_each_card_rtds(card, rtd) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int j; - - for_each_rtd_codec_dai(rtd, j, codec_dai) { - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } - - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); - } - - return ret; -} - /** * snd_soc_register_card - Register a card with the ASoC core * -- cgit From 74bd3f92d0d173fe4c0a12cf736c505ceb15576a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Nov 2019 10:08:06 +0900 Subject: ASoC: soc-core: tidyup soc_probe_aux_devices() snd_soc_bind_card() is calling many initialize functions for each card / link / dai / aux etc, etc, etc... When error happen, the message is indicated at snd_soc_bind_card(), not at each functions. But, only soc_probe_aux_devices() case is indicating error at functions, not at snd_soc_bind_card(). It is not an issue, but unbalanced. This patch moves error message to snd_soc_bind_card(). Also avoids deep-nested code. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87lfsthkw9.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f3d33a908fbc..92260a9569a2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1579,21 +1579,18 @@ static int soc_bind_aux_dev(struct snd_soc_card *card) static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp; + struct snd_soc_component *component; int order; int ret; for_each_comp_order(order) { - for_each_card_auxs(card, comp) { - if (comp->driver->probe_order == order) { - ret = soc_probe_component(card, comp); - if (ret < 0) { - dev_err(card->dev, - "ASoC: failed to probe aux component %s %d\n", - comp->name, ret); - return ret; - } - } + for_each_card_auxs(card, component) { + if (component->driver->probe_order != order) + continue; + + ret = soc_probe_component(card, component); + if (ret < 0) + return ret; } } @@ -2042,8 +2039,11 @@ static int snd_soc_bind_card(struct snd_soc_card *card) /* probe auxiliary components */ ret = soc_probe_aux_devices(card); - if (ret < 0) + if (ret < 0) { + dev_err(card->dev, + "ASoC: failed to probe aux component %d\n", ret); goto probe_end; + } /* probe all DAI links on this card */ ret = soc_probe_link_dais(card); -- cgit From e9149b8c00d25dbaef1aa174fc604bed207e576d Mon Sep 17 00:00:00 2001 From: Michael Walle Date: Tue, 12 Nov 2019 23:36:29 +0100 Subject: ASoC: wm8904: fix regcache handling The current code assumes that the power is turned off in SND_SOC_BIAS_OFF. If there are no actual regulator the codec isn't turned off and the registers are not reset to their default values but the regcache is still marked as dirty. Thus a value might not be written to the hardware if it is set to the default value. Do a software reset before turning off the power to make sure the registers are always reset to their default states. Signed-off-by: Michael Walle Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20191112223629.21867-1-michael@walle.cc Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index bcb3c9d5abf0..9e8c564f6e9c 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1917,6 +1917,7 @@ static int wm8904_set_bias_level(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); + snd_soc_component_write(component, WM8904_SW_RESET_AND_ID, 0); regcache_cache_only(wm8904->regmap, true); regcache_mark_dirty(wm8904->regmap); -- cgit From fe23be2d85b05f561431d75acddec726ea807d2a Mon Sep 17 00:00:00 2001 From: Jacob Rasmussen Date: Thu, 14 Nov 2019 16:20:11 -0700 Subject: ASoC: rt5645: Fixed typo for buddy jack support. Had a typo in e7cfd867fd98 that resulted in buddy jack support not being fixed. Fixes: e7cfd867fd98 ("ASoC: rt5645: Fixed buddy jack support.") Signed-off-by: Jacob Rasmussen Reviewed-by: Ross Zwisler Cc: CC: stable@vger.kernel.org Link: https://lore.kernel.org/r/20191114232011.165762-1-jacobraz@google.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 902ac98a3fbe..19662ee330d6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3271,7 +3271,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) report, SND_JACK_MICROPHONE); return; case 4: - val = snd_soc_component_read32(rt5645->component, RT5645_A_JD_CTRL1) & 0x002; + val = snd_soc_component_read32(rt5645->component, RT5645_A_JD_CTRL1) & 0x0020; break; default: /* read rt5645 jd1_1 status */ val = snd_soc_component_read32(rt5645->component, RT5645_INT_IRQ_ST) & 0x1000; -- cgit From b26eb5173c85082eec7d6e18369f6f9d96bf0b21 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 12 Nov 2019 19:02:18 +0000 Subject: ASoC: tas2770: clean up an indentation issue There is a block that is indented too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20191112190218.282337-1-colin.king@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index ad76f22fcfac..54c8135fe43c 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -761,12 +761,12 @@ static int tas2770_i2c_probe(struct i2c_client *client, tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, "reset-gpio", GPIOD_OUT_HIGH); - if (IS_ERR(tas2770->reset_gpio)) { - if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { - tas2770->reset_gpio = NULL; - return -EPROBE_DEFER; - } + if (IS_ERR(tas2770->reset_gpio)) { + if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { + tas2770->reset_gpio = NULL; + return -EPROBE_DEFER; } + } tas2770->channel_size = 0; tas2770->slot_width = 0; -- cgit From 2a6f0892bda954dc2688b002060093ee0fe38528 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 13 Nov 2019 10:16:29 +0900 Subject: ASoC: soc-core: call snd_soc_dapm_shutdown() at soc_cleanup_card_resources() It is easy to read code if it is cleanly using paired function/naming, like start <-> stop, register <-> unregister, etc, etc. But, current ALSA SoC code is very random, unbalance, not paired, etc. It is easy to create bug at the such code, and it will be difficult to debug. snd_soc_bind_card() is calling snd_soc_dapm_init() for both card and component. Let's call paired snd_soc_dapm_shutdown() at paired soc_cleanup_card_resources(). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87r22c4lub.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 92260a9569a2..216000ae3e20 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1953,6 +1953,8 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) card->snd_card = NULL; } + snd_soc_dapm_shutdown(card); + /* remove and free each DAI */ soc_remove_link_dais(card); @@ -2389,7 +2391,6 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) { if (card->instantiated) { card->instantiated = false; - snd_soc_dapm_shutdown(card); snd_soc_flush_all_delayed_work(card); soc_cleanup_card_resources(card); -- cgit From 2cc1afcfc617b33b72e409360ba5dd7c3fc2492d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 13 Nov 2019 10:16:34 +0900 Subject: ASoC: soc-core: move snd_soc_unbind_card() next to snd_soc_bind_card() To makes code readable, this patch moves snd_soc_unbind_card() next to snd_soc_bind_card(). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87pnhw4lu5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 216000ae3e20..a149697e705d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1973,6 +1973,21 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) card->remove(card); } +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_flush_all_delayed_work(card); + + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); + } +} + static int snd_soc_bind_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -2387,21 +2402,6 @@ int snd_soc_register_card(struct snd_soc_card *card) } EXPORT_SYMBOL_GPL(snd_soc_register_card); -static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) -{ - if (card->instantiated) { - card->instantiated = false; - snd_soc_flush_all_delayed_work(card); - - soc_cleanup_card_resources(card); - if (!unregister) - list_add(&card->list, &unbind_card_list); - } else { - if (unregister) - list_del(&card->list); - } -} - /** * snd_soc_unregister_card - Unregister a card with the ASoC core * -- cgit From ce21401c9d30e81824e864dc25824ebeb9690801 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 13 Nov 2019 10:16:48 +0900 Subject: ASoC: soc-core: care card_probed at soc_cleanup_card_resources() soc_cleanup_card_resources() will call card->remove(), but it should be called if card->probe() or card->late_probe() are called. snd_soc_bind_card() might be error before calling card->probe() / card->late_probe(). In that time, card->remove() will be called. This patch adds card_probed parameter to judge it. Fixes: bfce78a559655 ("ASoC: soc-core: tidyup soc_init_dai_link()") Signed-off-by: Kuninori Morimoto Tested-by: Jon Hunter Link: https://lore.kernel.org/r/87o8xg4ltr.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a149697e705d..977a7bfad519 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1940,7 +1940,8 @@ static void __soc_setup_card_name(char *name, int len, } } -static void soc_cleanup_card_resources(struct snd_soc_card *card) +static void soc_cleanup_card_resources(struct snd_soc_card *card, + int card_probed) { struct snd_soc_dai_link *link, *_link; @@ -1969,17 +1970,19 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) soc_cleanup_card_debugfs(card); /* remove the card */ - if (card->remove) + if (card_probed && card->remove) card->remove(card); } static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) { if (card->instantiated) { + int card_probed = 1; + card->instantiated = false; snd_soc_flush_all_delayed_work(card); - soc_cleanup_card_resources(card); + soc_cleanup_card_resources(card, card_probed); if (!unregister) list_add(&card->list, &unbind_card_list); } else { @@ -1992,7 +1995,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link *dai_link; - int ret, i; + int ret, i, card_probed = 0; mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); @@ -2044,6 +2047,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) ret = card->probe(card); if (ret < 0) goto probe_end; + card_probed = 1; } /* probe all components used by DAI links on this card */ @@ -2109,6 +2113,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) goto probe_end; } } + card_probed = 1; snd_soc_dapm_new_widgets(card); @@ -2138,7 +2143,7 @@ static int snd_soc_bind_card(struct snd_soc_card *card) probe_end: if (ret < 0) - soc_cleanup_card_resources(card); + soc_cleanup_card_resources(card, card_probed); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); -- cgit From de8cf95231102f0b1d71499b89c93902c9bb7908 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 13 Nov 2019 11:54:44 +0200 Subject: ASoC: dmaengine: Use dma_request_chan() directly for channel request dma_request_slave_channel_reason() is: #define dma_request_slave_channel_reason(dev, name) \ dma_request_chan(dev, name) Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20191113095445.3211-2-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index f4c755209e03..a428ff393ea2 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -387,7 +387,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = dmaengine_pcm_dma_channel_names[i]; if (config && config->chan_names[i]) name = config->chan_names[i]; - chan = dma_request_slave_channel_reason(dev, name); + chan = dma_request_chan(dev, name); if (IS_ERR(chan)) { if (PTR_ERR(chan) == -EPROBE_DEFER) return -EPROBE_DEFER; -- cgit From f0c97131b946fe77e22eddbd7e7d92fb6459f497 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 13 Nov 2019 11:54:45 +0200 Subject: ASoC: ti: davinci-mcasp: Use dma_request_chan() directly for channel request dma_request_slave_channel_reason() is: #define dma_request_slave_channel_reason(dev, name) \ dma_request_chan(dev, name) Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20191113095445.3211-3-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 7aa3c32e4a49..8e5371801d88 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1867,7 +1867,7 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) return PCM_EDMA; tmp = mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data; - chan = dma_request_slave_channel_reason(mcasp->dev, tmp); + chan = dma_request_chan(mcasp->dev, tmp); if (IS_ERR(chan)) { if (PTR_ERR(chan) != -EPROBE_DEFER) dev_err(mcasp->dev, -- cgit From 44d13f6c2a0b739a7c4df3c478c8070320c4fd45 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 18 Nov 2019 17:16:24 +0800 Subject: ASoC: rt5682: fix the charge pump capacitor discharges Due to switching the HV to LV mode while stopping playback, the charge pump capacitor will be discharged to the source of the pump circuit. Therefore, this patch removed the event control. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20191118091624.18699-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 27 +++------------------------ 1 file changed, 3 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 62b8ed412bd1..5370e4b00104 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1451,28 +1451,6 @@ static const struct snd_kcontrol_new hpor_switch = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5682_HP_CTRL_1, RT5682_R_MUTE_SFT, 1, 1); -static int rt5682_charge_pump_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_component *component = - snd_soc_dapm_to_component(w->dapm); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_update_bits(component, - RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_HV); - break; - case SND_SOC_DAPM_POST_PMD: - snd_soc_component_update_bits(component, - RT5682_HP_CHARGE_PUMP_1, RT5682_PM_HP_MASK, RT5682_PM_HP_LV); - break; - default: - return 0; - } - - return 0; -} - static int rt5682_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1756,8 +1734,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("HP Amp R", RT5682_PWR_ANLG_1, RT5682_PWR_HA_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5682_DEPOP_1, - RT5682_PUMP_EN_SFT, 0, rt5682_charge_pump_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + RT5682_PUMP_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5682_DEPOP_1, RT5682_CAPLESS_EN_SFT, 0, NULL, 0), @@ -2655,6 +2632,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); + regmap_update_bits(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, + RT5682_PM_HP_MASK, RT5682_PM_HP_HV); INIT_DELAYED_WORK(&rt5682->jack_detect_work, rt5682_jack_detect_handler); -- cgit From a3b9ed55775eb18ca9f1641f13328f479a3653cb Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Thu, 14 Nov 2019 23:33:04 +0800 Subject: ASoC: rt5677: rt5677_check_hotword() can be static Fixes: 21c00e5df439 ("ASoC: rt5677: Enable jack detect while DSP is running") Signed-off-by: kbuild test robot Link: https://lore.kernel.org/r/20191114153304.n4pyix7qadu76tx4@4978f4969bb8 Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2f763b4c399..e9a051a50ab2 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5307,7 +5307,7 @@ static const struct rt5677_irq_desc rt5677_irq_descs[] = { }, }; -bool rt5677_check_hotword(struct rt5677_priv *rt5677) +static bool rt5677_check_hotword(struct rt5677_priv *rt5677) { int reg_gpio; -- cgit From 51d2584a98942a6b4a0dfa183f4a9afcb8a04073 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 15 Nov 2019 15:57:04 +0530 Subject: ASoC: qcom: q6asm: add support to flac config Qualcomm DSPs expect flac config to be set for flac decoders, so add the API to program the flac config to the DSP Signed-off-by: Srinivas Kandagatla Signed-off-by: Vinod Koul Link: https://lore.kernel.org/r/20191115102705.649976-3-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm.c | 55 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 15 ++++++++++++ 2 files changed, 70 insertions(+) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index e8141a33a55e..36e0eab13a98 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -38,6 +38,7 @@ #define ASM_SESSION_CMD_RUN_V2 0x00010DAA #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 #define ASM_MEDIA_FMT_MP3 0x00010BE9 +#define ASM_MEDIA_FMT_FLAC 0x00010C16 #define ASM_DATA_CMD_WRITE_V2 0x00010DAB #define ASM_DATA_CMD_READ_V2 0x00010DAC #define ASM_SESSION_CMD_SUSPEND 0x00010DEC @@ -89,6 +90,20 @@ struct asm_multi_channel_pcm_fmt_blk_v2 { u8 channel_mapping[PCM_MAX_NUM_CHANNEL]; } __packed; +struct asm_flac_fmt_blk_v2 { + struct asm_data_cmd_media_fmt_update_v2 fmt_blk; + u16 is_stream_info_present; + u16 num_channels; + u16 min_blk_size; + u16 max_blk_size; + u16 md5_sum[8]; + u32 sample_rate; + u32 min_frame_size; + u32 max_frame_size; + u16 sample_size; + u16 reserved; +} __packed; + struct asm_stream_cmd_set_encdec_param { u32 param_id; u32 param_size; @@ -876,6 +891,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, case FORMAT_LINEAR_PCM: open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2; break; + case SND_AUDIOCODEC_FLAC: + open->dec_fmt_id = ASM_MEDIA_FMT_FLAC; + break; default: dev_err(ac->dev, "Invalid format 0x%x\n", format); rc = -EINVAL; @@ -1021,6 +1039,42 @@ err: } EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm); + +int q6asm_stream_media_format_block_flac(struct audio_client *ac, + struct q6asm_flac_cfg *cfg) +{ + struct asm_flac_fmt_blk_v2 *fmt; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(*fmt); + p = kzalloc(pkt_size, GFP_KERNEL); + if (!p) + return -ENOMEM; + + pkt = p; + fmt = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + + pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; + fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); + fmt->is_stream_info_present = cfg->stream_info_present; + fmt->num_channels = cfg->ch_cfg; + fmt->min_blk_size = cfg->min_blk_size; + fmt->max_blk_size = cfg->max_blk_size; + fmt->sample_rate = cfg->sample_rate; + fmt->min_frame_size = cfg->min_frame_size; + fmt->max_frame_size = cfg->max_frame_size; + fmt->sample_size = cfg->sample_size; + + rc = q6asm_ac_send_cmd_sync(ac, pkt); + kfree(pkt); + + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac); /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * @@ -1075,6 +1129,7 @@ err: } EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support); + /** * q6asm_read() - read data of period size from audio client * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 9f5fb573e4a0..6764f55f7078 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -32,6 +32,19 @@ enum { #define NO_TIMESTAMP 0xFF00 #define FORMAT_LINEAR_PCM 0x0000 +struct q6asm_flac_cfg { + u32 sample_rate; + u32 ext_sample_rate; + u32 min_frame_size; + u32 max_frame_size; + u16 stream_info_present; + u16 min_blk_size; + u16 max_blk_size; + u16 ch_cfg; + u16 sample_size; + u16 md5_sum; +}; + typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token, void *payload, void *priv); struct audio_client; @@ -54,6 +67,8 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, uint32_t rate, uint32_t channels, u8 channel_map[PCM_MAX_NUM_CHANNEL], uint16_t bits_per_sample); +int q6asm_stream_media_format_block_flac(struct audio_client *ac, + struct q6asm_flac_cfg *cfg); int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, -- cgit From baddcee989931e304eeb90c101751c2f7f9b5045 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 15 Nov 2019 15:57:05 +0530 Subject: ASoC: qcom: q6asm-dai: add support to flac decoder Qualcomm DSPs also support the flac decoder, so add support for FLAC decoder and convert the snd_dec_flac params to qdsp format. Co-developed-by: Srinivas Kandagatla Signed-off-by: Srinivas Kandagatla Signed-off-by: Vinod Koul Link: https://lore.kernel.org/r/20191115102705.649976-4-vkoul@kernel.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6asm-dai.c | 35 ++++++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index f59353f510b8..8150c10f081e 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -626,8 +626,14 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); int dir = stream->direction; struct q6asm_dai_data *pdata; + struct q6asm_flac_cfg flac_cfg; struct device *dev = c->dev; int ret; + union snd_codec_options *codec_options; + struct snd_dec_flac *flac; + + codec_options = &(prtd->codec_param.codec.options); + memcpy(&prtd->codec_param, params, sizeof(*params)); @@ -664,6 +670,32 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, return ret; } + switch (params->codec.id) { + case SND_AUDIOCODEC_FLAC: + + memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); + flac = &codec_options->flac_d; + + flac_cfg.ch_cfg = params->codec.ch_in; + flac_cfg.sample_rate = params->codec.sample_rate; + flac_cfg.stream_info_present = 1; + flac_cfg.sample_size = flac->sample_size; + flac_cfg.min_blk_size = flac->min_blk_size; + flac_cfg.max_blk_size = flac->max_blk_size; + flac_cfg.max_frame_size = flac->max_frame_size; + flac_cfg.min_frame_size = flac->min_frame_size; + + ret = q6asm_stream_media_format_block_flac(prtd->audio_client, + &flac_cfg); + if (ret < 0) { + dev_err(dev, "FLAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + default: + break; + } + ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, (prtd->pcm_size / prtd->periods), prtd->periods); @@ -759,8 +791,9 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; - caps->num_codecs = 1; + caps->num_codecs = 2; caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_FLAC; return 0; } -- cgit From 2dab09be49a1e7a4dd13cb47d3a1441a2ef33a87 Mon Sep 17 00:00:00 2001 From: Chuhong Yuan Date: Mon, 18 Nov 2019 15:36:33 +0800 Subject: ASoC: wm2200: add missed operations in remove and probe failure This driver misses calls to pm_runtime_disable and regulator_bulk_disable in remove and a call to free_irq in probe failure. Add the calls to fix it. Signed-off-by: Chuhong Yuan Link: https://lore.kernel.org/r/20191118073633.28237-1-hslester96@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index cf64e109c658..7b087d94141b 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2410,6 +2410,8 @@ static int wm2200_i2c_probe(struct i2c_client *i2c, err_pm_runtime: pm_runtime_disable(&i2c->dev); + if (i2c->irq) + free_irq(i2c->irq, wm2200); err_reset: if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); @@ -2426,12 +2428,15 @@ static int wm2200_i2c_remove(struct i2c_client *i2c) { struct wm2200_priv *wm2200 = i2c_get_clientdata(i2c); + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); if (wm2200->pdata.reset) gpio_set_value_cansleep(wm2200->pdata.reset, 0); if (wm2200->pdata.ldo_ena) gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); return 0; } -- cgit From b1176bbb70866f24099cd2720283c7219fb4a81c Mon Sep 17 00:00:00 2001 From: Chuhong Yuan Date: Mon, 18 Nov 2019 15:37:07 +0800 Subject: ASoC: wm5100: add missed pm_runtime_disable The driver forgets to call pm_runtime_disable in remove and probe failure. Add the calls to fix it. Signed-off-by: Chuhong Yuan Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20191118073707.28298-1-hslester96@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4af0e519e623..91cc63c5a51f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2617,6 +2617,7 @@ static int wm5100_i2c_probe(struct i2c_client *i2c, return ret; err_reset: + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); @@ -2640,6 +2641,7 @@ static int wm5100_i2c_remove(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); + pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); -- cgit From 79f6c108c87b470aacf25fc25a86f48694d03ae8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 13 Nov 2019 14:47:34 +0200 Subject: ASoC: pcm3168a: Add support for optional RST gpio handling In case the RST line is connected to a GPIO line it needs to be pulled high when the driver probes to be able to use the codec. Add support also for cases when more than one codec is is controlled by the same GPIO line by requesting the gpio with GPIOD_FLAGS_BIT_NONEXCLUSIVE. Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20191113124734.27984-3-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 38 ++++++++++++++++++++++++++++++++++---- 1 file changed, 34 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 313500ab36df..f3475134b519 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -9,7 +9,9 @@ #include #include +#include #include +#include #include #include @@ -59,6 +61,7 @@ struct pcm3168a_priv { struct regulator_bulk_data supplies[PCM3168A_NUM_SUPPLIES]; struct regmap *regmap; struct clk *scki; + struct gpio_desc *gpio_rst; unsigned long sysclk; struct pcm3168a_io_params io_params[2]; @@ -643,6 +646,7 @@ static bool pcm3168a_readable_register(struct device *dev, unsigned int reg) static bool pcm3168a_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { + case PCM3168A_RST_SMODE: case PCM3168A_DAC_ZERO: case PCM3168A_ADC_OV: return true; @@ -702,6 +706,21 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, pcm3168a); + /* + * Request the RST gpio line as non exclusive as the same reset line + * might be connected to multiple pcm3168a codec + */ + pcm3168a->gpio_rst = devm_gpiod_get_optional(dev, "rst", + GPIOD_OUT_HIGH | + GPIOD_FLAGS_BIT_NONEXCLUSIVE); + if (IS_ERR(pcm3168a->gpio_rst)) { + ret = PTR_ERR(pcm3168a->gpio_rst); + if (ret != -EPROBE_DEFER ) + dev_err(dev, "failed to acquire RST gpio: %d\n", ret); + + return ret; + } + pcm3168a->scki = devm_clk_get(dev, "scki"); if (IS_ERR(pcm3168a->scki)) { ret = PTR_ERR(pcm3168a->scki); @@ -743,10 +762,18 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) goto err_regulator; } - ret = pcm3168a_reset(pcm3168a); - if (ret) { - dev_err(dev, "Failed to reset device: %d\n", ret); - goto err_regulator; + if (pcm3168a->gpio_rst) { + /* + * The device is taken out from reset via GPIO line, wait for + * 3846 SCKI clock cycles for the internal reset de-assertion + */ + msleep(DIV_ROUND_UP(3846 * 1000, pcm3168a->sysclk)); + } else { + ret = pcm3168a_reset(pcm3168a); + if (ret) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err_regulator; + } } pm_runtime_set_active(dev); @@ -785,6 +812,9 @@ static void pcm3168a_disable(struct device *dev) void pcm3168a_remove(struct device *dev) { + struct pcm3168a_priv *pcm3168a = dev_get_drvdata(dev); + + gpiod_set_value_cansleep(pcm3168a->gpio_rst, 0); pm_runtime_disable(dev); #ifndef CONFIG_PM pcm3168a_disable(dev); -- cgit From 302ee055af0201e61be7670f957c1622d6ce176e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 18 Nov 2019 15:52:47 +0000 Subject: ASoC: SOF: Intel: Fix build break Commit 130d3e9077 (Fix CFL and CML FW nocodec binary names.) broke the build in some configurations as it depends on changes in the development branch, revert it. Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 2ef927371b23..d66412a77873 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -113,7 +113,7 @@ static const struct sof_dev_desc cnl_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) static const struct sof_dev_desc cfl_desc = { - .machines = snd_soc_acpi_intel_cfl_machines, + .machines = snd_soc_acpi_intel_cnl_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -122,7 +122,7 @@ static const struct sof_dev_desc cfl_desc = { .chip_info = &cnl_chip_info, .default_fw_path = "intel/sof", .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-cfl.ri", + .nocodec_fw_filename = "sof-cnl.ri", .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .arch_ops = &sof_xtensa_arch_ops @@ -133,7 +133,7 @@ static const struct sof_dev_desc cfl_desc = { IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) static const struct sof_dev_desc cml_desc = { - .machines = snd_soc_acpi_intel_cml_machines, + .machines = snd_soc_acpi_intel_cnl_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -142,7 +142,7 @@ static const struct sof_dev_desc cml_desc = { .chip_info = &cnl_chip_info, .default_fw_path = "intel/sof", .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-cml.ri", + .nocodec_fw_filename = "sof-cnl.ri", .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .arch_ops = &sof_xtensa_arch_ops -- cgit From 653c28afa76b45c570370c3c3a89975c68c5fc8e Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 11 Nov 2019 16:29:00 -0600 Subject: ASoC: SOF: Intel: Fix CFL and CML FW nocodec binary names. The manifest information is different between CNL, CML and CFL platforms hence we need to load different files. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191111222901.19892-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 4de90d04a91b..3252dbe277c8 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -119,7 +119,7 @@ static const struct sof_dev_desc cnl_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) static const struct sof_dev_desc cfl_desc = { - .machines = snd_soc_acpi_intel_cnl_machines, + .machines = snd_soc_acpi_intel_cfl_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -128,7 +128,7 @@ static const struct sof_dev_desc cfl_desc = { .chip_info = &cnl_chip_info, .default_fw_path = "intel/sof", .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-cnl.ri", + .nocodec_fw_filename = "sof-cfl.ri", .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .arch_ops = &sof_xtensa_arch_ops @@ -139,7 +139,7 @@ static const struct sof_dev_desc cfl_desc = { IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) static const struct sof_dev_desc cml_desc = { - .machines = snd_soc_acpi_intel_cnl_machines, + .machines = snd_soc_acpi_intel_cml_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -148,7 +148,7 @@ static const struct sof_dev_desc cml_desc = { .chip_info = &cnl_chip_info, .default_fw_path = "intel/sof", .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-cnl.ri", + .nocodec_fw_filename = "sof-cml.ri", .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .arch_ops = &sof_xtensa_arch_ops -- cgit From e48fdb53bd1fa50796b5a050e6e31fde3891a2c8 Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Mon, 18 Nov 2019 16:12:06 +0100 Subject: ASoC: tlv320aic31xx: configure output common-mode voltage The tlv320aic31xx devices allow to adjust the output common-mode voltage for best analog performance. The datasheet states that the common mode voltage should be set to be <= AVDD/2. This changes allows to configure the output common-mode voltage via a DT property. If the property is absent the voltage is automatically chosen as the highest voltage below/equal to AVDD/2. Signed-off-by: Lucas Stach Link: https://lore.kernel.org/r/20191118151207.28576-1-l.stach@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 45 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 8 +++++++ 2 files changed, 53 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index df627a08def9..f6f19fdc72f5 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -171,6 +171,7 @@ struct aic31xx_priv { int rate_div_line; bool master_dapm_route_applied; int irq; + u8 ocmv; /* output common-mode voltage */ }; struct aic31xx_rate_divs { @@ -1312,6 +1313,11 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) if (ret) return ret; + /* set output common-mode voltage */ + snd_soc_component_update_bits(component, AIC31XX_HPDRIVER, + AIC31XX_HPD_OCMV_MASK, + aic31xx->ocmv << AIC31XX_HPD_OCMV_SHIFT); + return 0; } @@ -1501,6 +1507,43 @@ exit: return IRQ_NONE; } +static void aic31xx_configure_ocmv(struct aic31xx_priv *priv) +{ + struct device *dev = priv->dev; + int dvdd, avdd; + u32 value; + + if (dev->fwnode && + fwnode_property_read_u32(dev->fwnode, "ai31xx-ocmv", &value)) { + /* OCMV setting is forced by DT */ + if (value <= 3) { + priv->ocmv = value; + return; + } + } + + avdd = regulator_get_voltage(priv->supplies[3].consumer); + dvdd = regulator_get_voltage(priv->supplies[5].consumer); + + if (avdd > 3600000 || dvdd > 1950000) { + dev_warn(dev, + "Too high supply voltage(s) AVDD: %d, DVDD: %d\n", + avdd, dvdd); + } else if (avdd == 3600000 && dvdd == 1950000) { + priv->ocmv = AIC31XX_HPD_OCMV_1_8V; + } else if (avdd >= 3300000 && dvdd >= 1800000) { + priv->ocmv = AIC31XX_HPD_OCMV_1_65V; + } else if (avdd >= 3000000 && dvdd >= 1650000) { + priv->ocmv = AIC31XX_HPD_OCMV_1_5V; + } else if (avdd >= 2700000 && dvdd >= 1525000) { + priv->ocmv = AIC31XX_HPD_OCMV_1_35V; + } else { + dev_warn(dev, + "Invalid supply voltage(s) AVDD: %d, DVDD: %d\n", + avdd, dvdd); + } +} + static int aic31xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1570,6 +1613,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } + aic31xx_configure_ocmv(aic31xx); + if (aic31xx->irq > 0) { regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1, AIC31XX_GPIO1_FUNC_MASK, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index cb024955c978..83a8c7604cc3 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -232,6 +232,14 @@ struct aic31xx_pdata { #define AIC31XX_HSD_HP 0x01 #define AIC31XX_HSD_HS 0x03 +/* AIC31XX_HPDRIVER */ +#define AIC31XX_HPD_OCMV_MASK GENMASK(4, 3) +#define AIC31XX_HPD_OCMV_SHIFT 3 +#define AIC31XX_HPD_OCMV_1_35V 0x0 +#define AIC31XX_HPD_OCMV_1_5V 0x1 +#define AIC31XX_HPD_OCMV_1_65V 0x2 +#define AIC31XX_HPD_OCMV_1_8V 0x3 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- cgit From eb65ccdb083639f8a9b6919c94d1df570396a9a1 Mon Sep 17 00:00:00 2001 From: Li Xu Date: Fri, 15 Nov 2019 13:54:13 -0600 Subject: ASoC: wm_adsp: Expose mixer control API Expose mixer control API for reading and writing controls from the kernel. This API can be used by ALSA kernel drivers with ADSP support to read and write firmware-defined memory regions. Signed-off-by: Li Xu Signed-off-by: David Rhodes Acked-by: Charles Keepax Link: https://lore.kernel.org/r/1573847653-17094-2-git-send-email-david.rhodes@cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 81 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 4 +++ 2 files changed, 84 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9b8bb7bbe945..2a9b610f6d43 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -599,6 +599,9 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; const char *fw_name; + /* Subname is needed to match with firmware */ + const char *subname; + unsigned int subname_len; struct wm_adsp_alg_region alg_region; struct wm_coeff_ctl_ops ops; struct wm_adsp *dsp; @@ -1399,6 +1402,7 @@ static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) { kfree(ctl->cache); kfree(ctl->name); + kfree(ctl->subname); kfree(ctl); } @@ -1472,6 +1476,15 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ret = -ENOMEM; goto err_ctl; } + if (subname) { + ctl->subname_len = subname_len; + ctl->subname = kmemdup(subname, + strlen(subname) + 1, GFP_KERNEL); + if (!ctl->subname) { + ret = -ENOMEM; + goto err_ctl_name; + } + } ctl->enabled = 1; ctl->set = 0; ctl->ops.xget = wm_coeff_get; @@ -1485,7 +1498,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { ret = -ENOMEM; - goto err_ctl_name; + goto err_ctl_subname; } list_add(&ctl->list, &dsp->ctl_list); @@ -1508,6 +1521,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, err_ctl_cache: kfree(ctl->cache); +err_ctl_subname: + kfree(ctl->subname); err_ctl_name: kfree(ctl->name); err_ctl: @@ -1995,6 +2010,70 @@ out: return ret; } +/* + * Find wm_coeff_ctl with input name as its subname + * If not found, return NULL + */ +static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, + const char *name, int type, + unsigned int alg) +{ + struct wm_coeff_ctl *pos, *rslt = NULL; + + list_for_each_entry(pos, &dsp->ctl_list, list) { + if (!pos->subname) + continue; + if (strncmp(pos->subname, name, pos->subname_len) == 0 && + pos->alg_region.alg == alg && + pos->alg_region.type == type) { + rslt = pos; + break; + } + } + + return rslt; +} + +int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, + unsigned int alg, void *buf, size_t len) +{ + struct wm_coeff_ctl *ctl; + struct snd_kcontrol *kcontrol; + int ret; + + ctl = wm_adsp_get_ctl(dsp, name, type, alg); + if (!ctl) + return -EINVAL; + + if (len > ctl->len) + return -EINVAL; + + ret = wm_coeff_write_control(ctl, buf, len); + + kcontrol = snd_soc_card_get_kcontrol(dsp->component->card, ctl->name); + snd_ctl_notify(dsp->component->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, &kcontrol->id); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); + +int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, + unsigned int alg, void *buf, size_t len) +{ + struct wm_coeff_ctl *ctl; + + ctl = wm_adsp_get_ctl(dsp, name, type, alg); + if (!ctl) + return -EINVAL; + + if (len > ctl->len) + return -EINVAL; + + return wm_coeff_read_control(ctl, buf, len); +} +EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); + static void wm_adsp_ctl_fixup_base(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region) { diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index aa634ef6c9f5..4c481cf20275 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -201,5 +201,9 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp); int wm_adsp_compr_copy(struct snd_compr_stream *stream, char __user *buf, size_t count); +int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, + unsigned int alg, void *buf, size_t len); +int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, + unsigned int alg, void *buf, size_t len); #endif -- cgit From 0bb887709eb16bdc4b5baddd8337abf3de72917f Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Tue, 19 Nov 2019 15:51:38 +0100 Subject: ASoC: Intel: bytcr_rt5640: Update quirk for Acer Switch 10 SW5-012 2-in-1 When the Acer Switch 10 SW5-012 quirk was added we did not have jack-detection support yet; and the builtin microphone selection of the original quirk is wrong too. Fix the microphone-input quirk and add jack-detection info so that the internal-microphone and headphone/set jack on the Switch 10 work properly. Signed-off-by: Hans de Goede Reviewed-by: Andy Shevchenko Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191119145138.59162-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9c1aa4ec9cba..dd2b5ad08659 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -405,10 +405,12 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW5-012"), }, - .driver_data = (void *)(BYT_RT5640_IN1_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), - + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { .matches = { -- cgit From b2b2afbb48eac7215f951a8a462aa6837e0d495f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 18 Nov 2019 10:50:32 +0900 Subject: ASoC: soc-component: tidyup snd_soc_pcm_component_new/free() parameter This patch uses rtd instead of pcm at snd_soc_pcm_component_new/free() parameter. This is prepare for dai_link remove bug fix on topology. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87pnhqx89j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-component.c | 8 +++----- sound/soc/soc-pcm.c | 4 ++-- 2 files changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 98ef0666add2..1590e805d016 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -498,9 +498,8 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream, return -EINVAL; } -int snd_soc_pcm_component_new(struct snd_pcm *pcm) +int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; int ret; @@ -516,13 +515,12 @@ int snd_soc_pcm_component_new(struct snd_pcm *pcm) return 0; } -void snd_soc_pcm_component_free(struct snd_pcm *pcm) +void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; for_each_rtd_components(rtd, rtdcom, component) if (component->driver->pcm_destruct) - component->driver->pcm_destruct(component, pcm); + component->driver->pcm_destruct(component, rtd->pcm); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4bf71e3211d8..c624d30bfa3c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2898,7 +2898,7 @@ static void soc_pcm_private_free(struct snd_pcm *pcm) /* need to sync the delayed work before releasing resources */ flush_delayed_work(&rtd->delayed_work); - snd_soc_pcm_component_free(pcm); + snd_soc_pcm_component_free(rtd); } /* create a new pcm */ @@ -3036,7 +3036,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - ret = snd_soc_pcm_component_new(pcm); + ret = snd_soc_pcm_component_new(rtd); if (ret < 0) { dev_err(rtd->dev, "ASoC: pcm constructor failed: %d\n", ret); return ret; -- cgit From 0ced7b050224b18ca73e38e7068f36be8e708c06 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 18 Nov 2019 10:51:11 +0900 Subject: ASoC: soc-pcm: remove soc_pcm_private_free() soc-topology adds extra dai_link by using snd_soc_add_dai_link(), and removes it by snd_soc_romove_dai_link(). This snd_soc_add/remove_dai_link() and/or its related functions are unbalanced before, and now, these are balance-uped. But, it finds the random operation issue, and it is reported by Pierre-Louis. When card was released, topology will call snd_soc_remove_dai_link() via (A). static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; /* This should be called before snd_card_free() */ (A) soc_remove_link_components(card); /* free the ALSA card at first; this syncs with pending operations */ if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } /* remove and free each DAI */ (X) soc_remove_link_dais(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... } At (A), topology calls snd_soc_remove_dai_link(). Then topology rtd, and its related all data are freed. Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free() is called. static void soc_pcm_private_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *rtd = pcm->private_data; /* need to sync the delayed work before releasing resources */ flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); } Here, it gets rtd via pcm->private_data. But, topology related rtd are already freed at (A). Normal sound card has no damage, becase it frees rtd at (C). These are finalizing rtd related data. Thus, these should be called when rtd was freed, not sound card was freed. It is very natural and understandable. In other words, pcm->private_free = soc_pcm_private_free() is no longer needed. Extra issue is that there is zero chance to call soc_remove_dai() for topology related dai at (X). Because (A) removes rtd connection from card too, and, (X) is based on card connected rtd. This means, (X) need to be called before (C) (= for normal sound) and (A) (= for topology). Now, I want to focus this patch which is the reason why snd_card_free() = (B) is located there. commit 4efda5f2130da033aeedc5b3205569893b910de2 ("ASoC: Fix use-after-free at card unregistration") Original snd_card_free() was called last of this function. But moved to top to avoid use-after-free issue. The issue was happen at soc_pcm_free() which was pcm->private_free, today it is updated/renamed to soc_pcm_private_free(). In other words, (B) need to be called before (C) (= for normal sound) and (A) (= for topology), because it needs (not yet freed) rtd. But, (A) need to be called before (B), because it needs card->snd_card pointer. If we call flush_delayed_work() and snd_soc_pcm_component_free() (= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)), there is no reason to call snd_card_free() at top of this function. It can be called end of this function, again. But, in such case, it will likely break unbind again, as Takashi-san reported. When unbind is performed in a busy state, the code may release still-in-use resources. At least we need to call snd_card_disconnect_sync() at the first place. The final code will be... static void soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_dai_link *link, *_link; if (card->snd_card) (Z) snd_card_disconnect_sync(card->snd_card); (X) soc_remove_link_dais(card); (A) soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) (C) snd_soc_remove_dai_link(card, link); ... if (card->snd_card) { (B) snd_card_free(card->snd_card); card->snd_card = NULL; } } To avoid release still-in-use resources, call snd_card_disconnect_sync() at (Z). (X) is needed for both non-topology and topology. topology removes rtd via (A), and non topology removes rtd via (C). snd_card_free() is no longer related to use-after-free issue. Thus, locating (B) is no problem. Fixes: df95a16d2a9626 ("ASoC: soc-core: fix RIP warning on card removal") Fixes: bc7a9091e5b927 ("ASoC: soc-core: add soc_unbind_dai_link()") Reported-by: Pierre-Louis Bossart Signed-off-by: Kuninori Morimoto Tested-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 19 +++++++++++-------- sound/soc/soc-pcm.c | 10 ---------- 2 files changed, 11 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 977a7bfad519..e3a53ef1db04 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -419,6 +419,9 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) list_del(&rtd->list); + flush_delayed_work(&rtd->delayed_work); + snd_soc_pcm_component_free(rtd); + /* * we don't need to call kfree() for rtd->dev * see @@ -1945,19 +1948,14 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card, { struct snd_soc_dai_link *link, *_link; - /* This should be called before snd_card_free() */ - soc_remove_link_components(card); - - /* free the ALSA card at first; this syncs with pending operations */ - if (card->snd_card) { - snd_card_free(card->snd_card); - card->snd_card = NULL; - } + if (card->snd_card) + snd_card_disconnect_sync(card->snd_card); snd_soc_dapm_shutdown(card); /* remove and free each DAI */ soc_remove_link_dais(card); + soc_remove_link_components(card); for_each_card_links_safe(card, link, _link) snd_soc_remove_dai_link(card, link); @@ -1972,6 +1970,11 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card, /* remove the card */ if (card_probed && card->remove) card->remove(card); + + if (card->snd_card) { + snd_card_free(card->snd_card); + card->snd_card = NULL; + } } static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c624d30bfa3c..2c4f50c44591 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2892,15 +2892,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } -static void soc_pcm_private_free(struct snd_pcm *pcm) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - - /* need to sync the delayed work before releasing resources */ - flush_delayed_work(&rtd->delayed_work); - snd_soc_pcm_component_free(rtd); -} - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -3042,7 +3033,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) return ret; } - pcm->private_free = soc_pcm_private_free; pcm->no_device_suspend = true; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", -- cgit From dc73d73aa7145f55412611f3eead1e85ae026785 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 19 Nov 2019 18:49:32 +0100 Subject: ASoC: add control components management This ASCII string can carry additional information about soundcard components or configuration. Add the possibility to set this string via the ASoC card. Signed-off-by: Jaroslav Kysela Cc: Mark Brown Link: https://lore.kernel.org/r/20191119174933.25526-1-perex@perex.cz Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e3a53ef1db04..cc0ef0fcc005 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2108,6 +2108,19 @@ static int snd_soc_bind_card(struct snd_soc_card *card) soc_setup_card_name(card->snd_card->driver, card->driver_name, card->name, 1); + if (card->components) { + /* the current implementation of snd_component_add() accepts */ + /* multiple components in the string separated by space, */ + /* but the string collision (identical string) check might */ + /* not work correctly */ + ret = snd_component_add(card->snd_card, card->components); + if (ret < 0) { + dev_err(card->dev, "ASoC: %s snd_component_add() failed: %d\n", + card->name, ret); + goto probe_end; + } + } + if (card->late_probe) { ret = card->late_probe(card); if (ret < 0) { -- cgit From 4ec48e7cbe6e70352c802b5cb172b00ebd8af8e0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 20 Nov 2019 15:17:53 +0200 Subject: ASoC: pcm3168a: Update the RST gpio handling to align with documentation The RST (reset-gpios) is low active so the driver must handle it accordingly. Add comments to explain clearly how the line is used. Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20191120131753.6831-3-peter.ujfalusi@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3168a.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index f3475134b519..9711fab296eb 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -707,11 +707,15 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) dev_set_drvdata(dev, pcm3168a); /* - * Request the RST gpio line as non exclusive as the same reset line - * might be connected to multiple pcm3168a codec + * Request the reset (connected to RST pin) gpio line as non exclusive + * as the same reset line might be connected to multiple pcm3168a codec + * + * The RST is low active, we want the GPIO line to be high initially, so + * request the initial level to LOW which in practice means DEASSERTED: + * The deasserted level of GPIO_ACTIVE_LOW is HIGH. */ - pcm3168a->gpio_rst = devm_gpiod_get_optional(dev, "rst", - GPIOD_OUT_HIGH | + pcm3168a->gpio_rst = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW | GPIOD_FLAGS_BIT_NONEXCLUSIVE); if (IS_ERR(pcm3168a->gpio_rst)) { ret = PTR_ERR(pcm3168a->gpio_rst); @@ -814,7 +818,13 @@ void pcm3168a_remove(struct device *dev) { struct pcm3168a_priv *pcm3168a = dev_get_drvdata(dev); - gpiod_set_value_cansleep(pcm3168a->gpio_rst, 0); + /* + * The RST is low active, we want the GPIO line to be low when the + * driver is removed, so set level to 1 which in practice means + * ASSERTED: + * The asserted level of GPIO_ACTIVE_LOW is LOW. + */ + gpiod_set_value_cansleep(pcm3168a->gpio_rst, 1); pm_runtime_disable(dev); #ifndef CONFIG_PM pcm3168a_disable(dev); -- cgit From 5cca59516de5df9de6bdecb328dd55fb5bcccb41 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Tue, 12 Nov 2019 18:46:42 +0800 Subject: ASoC: soc-pcm: check symmetry before hw_params This reverts commit 957ce0c6b8a1f (ASoC: soc-pcm: check symmetry after hw_params). That commit cause soc_pcm_params_symmetry can't take effect. cpu_dai->rate, cpu_dai->channels and cpu_dai->sample_bits are updated in the middle of soc_pcm_hw_params, so move soc_pcm_params_symmetry to the end of soc_pcm_hw_params is not a good solution, for judgement of symmetry in the function is always true. FIXME: According to the comments of that commit, I think the case described in the commit should disable symmetric_rates in Back-End, rather than changing the position of soc_pcm_params_symmetry. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1573555602-5403-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c4f50c44591..01eb8700c3de 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -861,6 +861,11 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, int i, ret = 0; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto out; + if (rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { @@ -940,9 +945,6 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; out: mutex_unlock(&rtd->card->pcm_mutex); return ret; -- cgit From 3efd72330543da44e82e9371dfb639802c886f6c Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 20 Nov 2019 21:32:52 +0800 Subject: ASoC: Fix Kconfig indentation Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191120133252.6365-1-krzk@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 24 ++++++++++++------------ sound/soc/sof/intel/Kconfig | 10 +++++----- 2 files changed, 17 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index dfa2c365379f..6c9fd9ad566e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -441,18 +441,18 @@ config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH If unsure select "N". config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH - tristate "CML with RT1011 and RT5682 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_RT1011 - select SND_SOC_RT5682 - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC machine driver for SOF platform with - RT1011 + RT5682 I2S codec. - Say Y if you have such a device. - If unsure select "N". + tristate "CML with RT1011 and RT5682 in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT1011 + select SND_SOC_RT5682 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC machine driver for SOF platform with + RT1011 + RT5682 I2S codec. + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 04d4929cf91f..92f7485b6994 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -264,16 +264,16 @@ config SND_SOC_SOF_ELKHARTLAKE config SND_SOC_SOF_JASPERLAKE_SUPPORT bool "SOF support for JasperLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the JasperLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the JasperLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_JASPERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON -- cgit From ef8e14794308a428b194f8b06ad9ae06b43466e4 Mon Sep 17 00:00:00 2001 From: Nilkanth Ahirrao Date: Thu, 21 Nov 2019 12:10:23 +0100 Subject: ASoC: rsnd: fix DALIGN register for SSIU MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The current driver only sets 0x76543210 and 0x67452301 for DALIGN. This doesn’t work well for TDM split and ex-split mode for all SSIU. This patch programs the DALIGN registers based on the SSIU number. Cc: Kuninori Morimoto Cc: Jiada Wang Cc: Andrew Gabbasov Fixes: a914e44693d41b ("ASoC: rsnd: more clear rsnd_get_dalign() for DALIGN") Signed-off-by: Nilkanth Ahirrao Signed-off-by: Eugeniu Rosca Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20191121111023.10976-1-erosca@de.adit-jv.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e9596c2096cd..a6c1cf987e6e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -376,6 +376,17 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) */ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { + static const u32 dalign_values[8][2] = { + {0x76543210, 0x67452301}, + {0x00000032, 0x00000023}, + {0x00007654, 0x00006745}, + {0x00000076, 0x00000067}, + {0xfedcba98, 0xefcdab89}, + {0x000000ba, 0x000000ab}, + {0x0000fedc, 0x0000efcd}, + {0x000000fe, 0x000000ef}, + }; + int id = 0, inv; struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); @@ -411,13 +422,18 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) target = cmd ? cmd : ssiu; } + if (mod == ssiu) + id = rsnd_mod_id_sub(mod); + /* Non target mod or non 16bit needs normal DALIGN */ if ((snd_pcm_format_width(runtime->format) != 16) || (mod != target)) - return 0x76543210; + inv = 0; /* Target mod needs inverted DALIGN when 16bit */ else - return 0x67452301; + inv = 1; + + return dalign_values[id][inv]; } u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) -- cgit From 4e01e5dbba96f731119f3f1a6bf51b54c98c5940 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 20 Nov 2019 18:44:34 +0100 Subject: ASoC: improve the DMI long card code in asoc-core Add append_dmi_string() function and make the code more readable. Signed-off-by: Jaroslav Kysela Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191120174435.30920-1-perex@perex.cz Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 66 ++++++++++++++++++++-------------------------------- 1 file changed, 25 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cc0ef0fcc005..a1f4d64a0a18 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1718,6 +1718,23 @@ static int is_dmi_valid(const char *field) return 1; } +/* + * Append a string to card->dmi_longname with character cleanups. + */ +static void append_dmi_string(struct snd_soc_card *card, const char *str) +{ + char *dst = card->dmi_longname; + size_t dst_len = sizeof(card->dmi_longname); + size_t len; + + len = strlen(dst); + snprintf(dst + len, dst_len - len, "-%s", str); + + len++; /* skip the separator "-" */ + if (len < dst_len) + cleanup_dmi_name(dst + len); +} + /** * snd_soc_set_dmi_name() - Register DMI names to card * @card: The card to register DMI names @@ -1752,61 +1769,36 @@ static int is_dmi_valid(const char *field) int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) { const char *vendor, *product, *product_version, *board; - size_t longname_buf_size = sizeof(card->snd_card->longname); - size_t len; if (card->long_name) return 0; /* long name already set by driver or from DMI */ - /* make up dmi long name as: vendor.product.version.board */ + /* make up dmi long name as: vendor-product-version-board */ vendor = dmi_get_system_info(DMI_BOARD_VENDOR); if (!vendor || !is_dmi_valid(vendor)) { dev_warn(card->dev, "ASoC: no DMI vendor name!\n"); return 0; } - snprintf(card->dmi_longname, sizeof(card->snd_card->longname), - "%s", vendor); + snprintf(card->dmi_longname, sizeof(card->dmi_longname), "%s", vendor); cleanup_dmi_name(card->dmi_longname); product = dmi_get_system_info(DMI_PRODUCT_NAME); if (product && is_dmi_valid(product)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", product); - - len++; /* skip the separator "-" */ - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); + append_dmi_string(card, product); /* * some vendors like Lenovo may only put a self-explanatory * name in the product version field */ product_version = dmi_get_system_info(DMI_PRODUCT_VERSION); - if (product_version && is_dmi_valid(product_version)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", product_version); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); - } + if (product_version && is_dmi_valid(product_version)) + append_dmi_string(card, product_version); } board = dmi_get_system_info(DMI_BOARD_NAME); if (board && is_dmi_valid(board)) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", board); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); + append_dmi_string(card, board); } else if (!product) { /* fall back to using legacy name */ dev_warn(card->dev, "ASoC: no DMI board/product name!\n"); @@ -1814,16 +1806,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) } /* Add flavour to dmi long name */ - if (flavour) { - len = strlen(card->dmi_longname); - snprintf(card->dmi_longname + len, - longname_buf_size - len, - "-%s", flavour); - - len++; - if (len < longname_buf_size) - cleanup_dmi_name(card->dmi_longname + len); - } + if (flavour) + append_dmi_string(card, flavour); /* set the card long name */ card->long_name = card->dmi_longname; -- cgit From 39870b0dec68ed7dd814beb697e541670975c7d8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 20 Nov 2019 18:44:35 +0100 Subject: ASoC: DMI long name - avoid to add board name if matches with product name Current code: LENOVO-20QE000VMC-ThinkPadX1Carbon7th-20QE000VMC With the patch: LENOVO-20QE000VMC-ThinkPadX1Carbon7th Signed-off-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20191120174435.30920-2-perex@perex.cz Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a1f4d64a0a18..062653ab03a3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1798,7 +1798,8 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) board = dmi_get_system_info(DMI_BOARD_NAME); if (board && is_dmi_valid(board)) { - append_dmi_string(card, board); + if (!product || strcasecmp(board, product)) + append_dmi_string(card, board); } else if (!product) { /* fall back to using legacy name */ dev_warn(card->dev, "ASoC: no DMI board/product name!\n"); -- cgit