diff options
30 files changed, 256 insertions, 300 deletions
diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7..e665f111b0d2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -985,6 +985,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* + * This driver uses legacy platform naming. Set by the core, machine + * drivers should not modify this value. + */ + unsigned int legacy_platform:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a5b09e75e787..f7d2b373da0a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -541,7 +541,8 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments == 0) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f48efce937ad..5957aeb1099e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2112,6 +2112,13 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, return 0; } +/* allow waiting for a capture stream that hasn't been started */ +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +#define wait_capture_start(substream) ((substream)->oss.oss) +#else +#define wait_capture_start(substream) false +#endif + /* the common loop for read/write data */ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, void *data, bool interleaved, @@ -2182,7 +2189,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, err = snd_pcm_start(substream); if (err < 0) goto _end_unlock; - } else { + } else if (!wait_capture_start(substream)) { /* nothing to do */ err = 0; goto _end_unlock; diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 598d140bb7cb..5fc497c6d738 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -903,6 +903,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; + if (!ins) + return 0; + snd_info_free_entry(ins->proc_sym_info_entry); ins->proc_sym_info_entry = NULL; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 51cc6589443f..152f54137082 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -931,6 +931,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 396ec43a2a54..b4f472157ebd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4102,6 +4102,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: + alc_process_coef_fw(codec, alc225_pre_hsmode); alc_process_coef_fw(codec, coef0225); break; case 0x10ec0867: @@ -5440,6 +5441,13 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, } } +static void alc_fixup_disable_mic_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5549,6 +5557,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC280_FIXUP_HP_HEADSET_MIC, @@ -6268,6 +6277,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DISABLE_MIC_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_mic_vref, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -6277,7 +6292,7 @@ static const struct hda_fixup alc269_fixups[] = { {} }, .chained = true, - .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + .chain_id = ALC225_FIXUP_DISABLE_MIC_VREF }, [ALC280_FIXUP_HP_HEADSET_MIC] = { .type = HDA_FIXUP_FUNC, @@ -6911,7 +6926,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"}, {.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"}, {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, - {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc255-dell1"}, + {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 022a8912c8a2..3d58338fa3cf 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -611,14 +611,16 @@ static int acp3x_audio_probe(struct platform_device *pdev) } irqflags = *((unsigned int *)(pdev->dev.platform_data)); - adata = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dev_data), - GFP_KERNEL); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); return -ENODEV; } + adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); + if (!adata) + return -ENOMEM; + adata->acp3x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3ab2949c1dfa..b19d7a3e7a2c 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1890,51 +1890,31 @@ static void hdmi_codec_remove(struct snd_soc_component *component) pm_runtime_disable(&hdev->dev); } -#ifdef CONFIG_PM -static int hdmi_codec_prepare(struct device *dev) -{ - struct hdac_device *hdev = dev_to_hdac_dev(dev); - - pm_runtime_get_sync(&hdev->dev); - - /* - * Power down afg. - * codec_read is preferred over codec_write to set the power state. - * This way verb is send to set the power state and response - * is received. So setting power state is ensured without using loop - * to read the state. - */ - snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - - return 0; -} - -static void hdmi_codec_complete(struct device *dev) +#ifdef CONFIG_PM_SLEEP +static int hdmi_codec_resume(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + int ret; - /* Power up afg */ - snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - - hdac_hdmi_skl_enable_all_pins(hdev); - hdac_hdmi_skl_enable_dp12(hdev); - + ret = pm_runtime_force_resume(dev); + if (ret < 0) + return ret; /* * As the ELD notify callback request is not entertained while the * device is in suspend state. Need to manually check detection of * all pins here. pin capablity change is not support, so use the * already set pin caps. + * + * NOTE: this is safe to call even if the codec doesn't actually resume. + * The pin check involves only with DRM audio component hooks, so it + * works even if the HD-audio side is still dreaming peacefully. */ hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - - pm_runtime_put_sync(&hdev->dev); + return 0; } #else -#define hdmi_codec_prepare NULL -#define hdmi_codec_complete NULL +#define hdmi_codec_resume NULL #endif static const struct snd_soc_component_driver hdmi_hda_codec = { @@ -2135,75 +2115,6 @@ static int hdac_hdmi_dev_remove(struct hdac_device *hdev) } #ifdef CONFIG_PM -/* - * Power management sequences - * ========================== - * - * The following explains the PM handling of HDAC HDMI with its parent - * device SKL and display power usage - * - * Probe - * ----- - * In SKL probe, - * 1. skl_probe_work() powers up the display (refcount++ -> 1) - * 2. enumerates the codecs on the link - * 3. powers down the display (refcount-- -> 0) - * - * In HDAC HDMI probe, - * 1. hdac_hdmi_dev_probe() powers up the display (refcount++ -> 1) - * 2. probe the codec - * 3. put the HDAC HDMI device to runtime suspend - * 4. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * - * Once children are runtime suspended, SKL device also goes to runtime - * suspend - * - * HDMI Playback - * ------------- - * Open HDMI device, - * 1. skl_runtime_resume() invoked - * 2. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) - * - * Close HDMI device, - * 1. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * 2. skl_runtime_suspend() invoked - * - * S0/S3 Cycle with playback in progress - * ------------------------------------- - * When the device is opened for playback, the device is runtime active - * already and the display refcount is 1 as explained above. - * - * Entering to S3, - * 1. hdmi_codec_prepare() invoke the runtime resume of codec which just - * increments the PM runtime usage count of the codec since the device - * is in use already - * 2. skl_suspend() powers down the display (refcount-- -> 0) - * - * Wakeup from S3, - * 1. skl_resume() powers up the display (refcount++ -> 1) - * 2. hdmi_codec_complete() invokes the runtime suspend of codec which just - * decrements the PM runtime usage count of the codec since the device - * is in use already - * - * Once playback is stopped, the display refcount is set to 0 as explained - * above in the HDMI playback sequence. The PM handlings are designed in - * such way that to balance the refcount of display power when the codec - * device put to S3 while playback is going on. - * - * S0/S3 Cycle without playback in progress - * ---------------------------------------- - * Entering to S3, - * 1. hdmi_codec_prepare() invoke the runtime resume of codec - * 2. skl_runtime_resume() invoked - * 3. hdac_hdmi_runtime_resume() powers up the display (refcount++ -> 1) - * 4. skl_suspend() powers down the display (refcount-- -> 0) - * - * Wakeup from S3, - * 1. skl_resume() powers up the display (refcount++ -> 1) - * 2. hdmi_codec_complete() invokes the runtime suspend of codec - * 3. hdac_hdmi_runtime_suspend() powers down the display (refcount-- -> 0) - * 4. skl_runtime_suspend() invoked - */ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); @@ -2277,8 +2188,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) static const struct dev_pm_ops hdac_hdmi_pm = { SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL) - .prepare = hdmi_codec_prepare, - .complete = hdmi_codec_complete, + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, hdmi_codec_resume) }; static const struct hda_device_id hdmi_list[] = { diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 6cb1653be804..4cc24a5d5c31 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1400,24 +1400,20 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) if (ret != 0) { dev_err(component->dev, "Failed to set digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, PCM512x_ANALOG_MUTE_DET, mute_det, (mute_det & 0x3) == 0, 200, 10000); - - mutex_unlock(&pcm512x->mutex); } else { pcm512x->mute &= ~0x1; ret = pcm512x_update_mute(pcm512x); if (ret != 0) { dev_err(component->dev, "Failed to update digital mute: %d\n", ret); - mutex_unlock(&pcm512x->mutex); - return ret; + goto unlock; } regmap_read_poll_timeout(pcm512x->regmap, @@ -1428,9 +1424,10 @@ static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) 200, 10000); } +unlock: mutex_unlock(&pcm512x->mutex); - return 0; + return ret; } static const struct snd_soc_dai_ops pcm512x_dai_ops = { diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 0ef966d56bac..e2855ab9a2c6 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1128,8 +1128,11 @@ static int rt274_i2c_probe(struct i2c_client *i2c, return ret; } - regmap_read(rt274->regmap, + ret = regmap_read(rt274->regmap, RT274_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + if (ret) + return ret; + if (val != RT274_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %#x is not rt274\n", val); diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 4d46f4567c3a..bec2eefa8b0f 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -280,6 +280,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_component *component) rt5514_dsp = devm_kzalloc(component->dev, sizeof(*rt5514_dsp), GFP_KERNEL); + if (!rt5514_dsp) + return -ENOMEM; rt5514_dsp->dev = &rt5514_spi->dev; mutex_init(&rt5514_dsp->dma_lock); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 34cfaf8f6f34..89c43b26c379 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2512,6 +2512,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); mutex_unlock(&rt5682->calibrate_mutex); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d82a8301fd74..96944cff0ed7 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -849,18 +849,18 @@ #define RT5682_SCLK_SRC_PLL2 (0x2 << 13) #define RT5682_SCLK_SRC_SDW (0x3 << 13) #define RT5682_SCLK_SRC_RCCLK (0x4 << 13) -#define RT5682_PLL1_SRC_MASK (0x3 << 10) -#define RT5682_PLL1_SRC_SFT 10 -#define RT5682_PLL1_SRC_MCLK (0x0 << 10) -#define RT5682_PLL1_SRC_BCLK1 (0x1 << 10) -#define RT5682_PLL1_SRC_SDW (0x2 << 10) -#define RT5682_PLL1_SRC_RC (0x3 << 10) -#define RT5682_PLL2_SRC_MASK (0x3 << 8) -#define RT5682_PLL2_SRC_SFT 8 -#define RT5682_PLL2_SRC_MCLK (0x0 << 8) -#define RT5682_PLL2_SRC_BCLK1 (0x1 << 8) -#define RT5682_PLL2_SRC_SDW (0x2 << 8) -#define RT5682_PLL2_SRC_RC (0x3 << 8) +#define RT5682_PLL2_SRC_MASK (0x3 << 10) +#define RT5682_PLL2_SRC_SFT 10 +#define RT5682_PLL2_SRC_MCLK (0x0 << 10) +#define RT5682_PLL2_SRC_BCLK1 (0x1 << 10) +#define RT5682_PLL2_SRC_SDW (0x2 << 10) +#define RT5682_PLL2_SRC_RC (0x3 << 10) +#define RT5682_PLL1_SRC_MASK (0x3 << 8) +#define RT5682_PLL1_SRC_SFT 8 +#define RT5682_PLL1_SRC_MCLK (0x0 << 8) +#define RT5682_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5682_PLL1_SRC_SDW (0x2 << 8) +#define RT5682_PLL1_SRC_RC (0x3 << 8) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e2b5a11b16d1..f03195d2ab2e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -822,6 +822,10 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: + /* Initial cold start */ + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) + break; + /* Switch off BCLK_N Divider */ snd_soc_component_update_bits(component, AIC32X4_BCLKN, AIC32X4_BCLKEN, 0); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 392d5eef356d..99e07b01a2ce 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 99a62ba409df..bd9fd2035c55 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index afc559866095..91a2436ce952 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + int ret; + + ret = + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (ret) + return ret; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); return 0; } diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 68e6543e6cb0..99f2a0156ae8 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -192,7 +192,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c74c4f17316f..8f83b182c4f9 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -55,39 +55,6 @@ enum { GLK_DPCM_AUDIO_HDMI3_PB, }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop sysclk: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); - if (ret < 0) { - dev_err(card->dev, "can't set codec pll: %d\n", ret); - return ret; - } - } - - if (ret) - dev_err(card->dev, "failed to start internal clk: %d\n", ret); - - return ret; -} - static const struct snd_kcontrol_new geminilake_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = { { "Spk", NULL, "Speaker" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, /* digital mics */ @@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_jack *jack; int ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + /* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index eab1f439dd3f..a4022983a7ce 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -146,7 +146,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 60c94836bf5b..4ed5b7e17d44 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -336,9 +336,6 @@ static int skl_suspend(struct device *dev) skl->skl_sst->fw_loaded = false; } - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - return 0; } @@ -350,10 +347,6 @@ static int skl_resume(struct device *dev) struct hdac_ext_link *hlink = NULL; int ret; - /* Turned OFF in HDMI codec driver after codec reconfiguration */ - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); - /* * resume only when we are not in suspend active, otherwise need to * restore the device @@ -446,8 +439,10 @@ static int skl_free(struct hdac_bus *bus) snd_hdac_ext_bus_exit(bus); cancel_work_sync(&skl->probe_work); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); snd_hdac_i915_exit(bus); + } return 0; } @@ -814,7 +809,7 @@ static void skl_probe_work(struct work_struct *work) err = skl_platform_register(bus->dev); if (err < 0) { dev_err(bus->dev, "platform register failed: %d\n", err); - return; + goto out_err; } err = skl_machine_device_register(skl); diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 5b986b74dd36..548eb4fa2da6 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -570,10 +570,10 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)compress_event_handler, prtd, stream_id, LEGACY_PCM_MODE); - if (!prtd->audio_client) { + if (IS_ERR(prtd->audio_client)) { dev_err(dev, "Could not allocate memory\n"); - kfree(prtd); - return -ENOMEM; + ret = PTR_ERR(prtd->audio_client); + goto free_prtd; } size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * @@ -582,7 +582,7 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) &prtd->dma_buffer); if (ret) { dev_err(dev, "Cannot allocate buffer(s)\n"); - return ret; + goto free_client; } if (pdata->sid < 0) @@ -595,6 +595,13 @@ static int q6asm_dai_compr_open(struct snd_compr_stream *stream) runtime->private_data = prtd; return 0; + +free_client: + q6asm_audio_client_free(prtd->audio_client); +free_prtd: + kfree(prtd); + + return ret; } static int q6asm_dai_compr_free(struct snd_compr_stream *stream) @@ -874,7 +881,7 @@ static int of_q6asm_parse_dai_data(struct device *dev, for_each_child_of_node(dev->of_node, node) { ret = of_property_read_u32(node, "reg", &id); - if (ret || id > MAX_SESSIONS || id < 0) { + if (ret || id >= MAX_SESSIONS || id < 0) { dev_err(dev, "valid dai id not found:%d\n", ret); continue; } diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 1db8ef668223..6f66a58e23ca 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -158,17 +158,24 @@ static int sdm845_snd_hw_params(struct snd_pcm_substream *substream, return ret; } +static void sdm845_jack_free(struct snd_jack *jack) +{ + struct snd_soc_component *component = jack->private_data; + + snd_soc_component_set_jack(component, NULL, NULL); +} + static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); - int i, rval; + struct snd_jack *jack; + int rval; if (!pdata->jack_setup) { - struct snd_jack *jack; - rval = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_HEADPHONE | @@ -190,16 +197,22 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + jack = pdata->jack.jack; + component = codec_dai->component; - component = dai->component; - rval = snd_soc_component_set_jack( - component, &pdata->jack, NULL); + jack->private_data = component; + jack->private_free = sdm845_jack_free; + rval = snd_soc_component_set_jack(component, + &pdata->jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { dev_warn(card->dev, "Failed to set jack: %d\n", rval); return rval; } + break; + default: + break; } return 0; diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..aae450ba4f08 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -742,7 +742,7 @@ static struct snd_soc_component *soc_find_component( if (of_node) { if (component->dev->of_node == of_node) return component; - } else if (strcmp(component->name, name) == 0) { + } else if (name && strcmp(component->name, name) == 0) { return component; } } @@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { + if (!platform || dai_link->legacy_platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); if (!platform) return -ENOMEM; - dai_link->platform = platform; - platform->name = dai_link->platform_name; - platform->of_node = dai_link->platform_of_node; - platform->dai_name = NULL; + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; } /* if there's no platform we match on the empty platform */ @@ -1129,6 +1130,15 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if ((link->platform->of_node || link->platform->name) && + !soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1150,15 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if ((link->cpu_of_node || link->cpu_name) && + !soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified @@ -2739,15 +2758,18 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + mutex_lock(&client_mutex); for_each_card_prelinks(card, i, link) { ret = soc_init_dai_link(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + mutex_unlock(&client_mutex); return ret; } } + mutex_unlock(&client_mutex); dev_set_drvdata(card->dev, card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a5178845065b..2c4c13419539 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2019,19 +2019,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -2044,7 +2044,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index eeda6d5565bc..a10fcb5963c6 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -108,7 +108,7 @@ struct davinci_mcasp { /* Used for comstraint setting on the second stream */ u32 channels; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM struct davinci_mcasp_context context; #endif @@ -1486,74 +1486,6 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) return 0; } -#ifdef CONFIG_PM_SLEEP -static int davinci_mcasp_suspend(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - context->pm_state = pm_runtime_active(mcasp->dev); - if (!context->pm_state) - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); - } - - for (i = 0; i < mcasp->num_serializer; i++) - context->xrsr_regs[i] = mcasp_get_reg(mcasp, - DAVINCI_MCASP_XRSRCTL_REG(i)); - - pm_runtime_put_sync(mcasp->dev); - - return 0; -} - -static int davinci_mcasp_resume(struct snd_soc_dai *dai) -{ - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - struct davinci_mcasp_context *context = &mcasp->context; - u32 reg; - int i; - - pm_runtime_get_sync(mcasp->dev); - - for (i = 0; i < ARRAY_SIZE(context_regs); i++) - mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); - - if (mcasp->txnumevt) { - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); - } - if (mcasp->rxnumevt) { - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); - } - - for (i = 0; i < mcasp->num_serializer; i++) - mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), - context->xrsr_regs[i]); - - if (!context->pm_state) - pm_runtime_put_sync(mcasp->dev); - - return 0; -} -#else -#define davinci_mcasp_suspend NULL -#define davinci_mcasp_resume NULL -#endif - #define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -1571,8 +1503,6 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", .probe = davinci_mcasp_dai_probe, - .suspend = davinci_mcasp_suspend, - .resume = davinci_mcasp_resume, .playback = { .channels_min = 1, .channels_max = 32 * 16, @@ -1976,7 +1906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } mcasp->num_serializer = pdata->num_serializer; -#ifdef CONFIG_PM_SLEEP +#ifdef CONFIG_PM mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev, mcasp->num_serializer, sizeof(u32), GFP_KERNEL); @@ -2196,11 +2126,73 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int davinci_mcasp_runtime_suspend(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + context->afifo_regs[0] = mcasp_get_reg(mcasp, reg); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + context->afifo_regs[1] = mcasp_get_reg(mcasp, reg); + } + + for (i = 0; i < mcasp->num_serializer; i++) + context->xrsr_regs[i] = mcasp_get_reg(mcasp, + DAVINCI_MCASP_XRSRCTL_REG(i)); + + return 0; +} + +static int davinci_mcasp_runtime_resume(struct device *dev) +{ + struct davinci_mcasp *mcasp = dev_get_drvdata(dev); + struct davinci_mcasp_context *context = &mcasp->context; + u32 reg; + int i; + + for (i = 0; i < ARRAY_SIZE(context_regs); i++) + mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); + + if (mcasp->txnumevt) { + reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[0]); + } + if (mcasp->rxnumevt) { + reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + mcasp_set_reg(mcasp, reg, context->afifo_regs[1]); + } + + for (i = 0; i < mcasp->num_serializer; i++) + mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), + context->xrsr_regs[i]); + + return 0; +} + +#endif + +static const struct dev_pm_ops davinci_mcasp_pm_ops = { + SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend, + davinci_mcasp_runtime_resume, + NULL) +}; + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 25e287feb58c..723a583a8d57 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -1,5 +1,5 @@ config SND_SOC_XILINX_I2S - tristate "Audio support for the the Xilinx I2S" + tristate "Audio support for the Xilinx I2S" help Select this option to enable Xilinx I2S Audio. This enables I2S playback and capture using xilinx soft IP. In transmitter diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index d4ae9eff41ce..8b353166ad44 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -1,12 +1,11 @@ // SPDX-License-Identifier: GPL-2.0 -/* - * Xilinx ASoC I2S audio support - * - * Copyright (C) 2018 Xilinx, Inc. - * - * Author: Praveen Vuppala <praveenv@xilinx.com> - * Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com> - */ +// +// Xilinx ASoC I2S audio support +// +// Copyright (C) 2018 Xilinx, Inc. +// +// Author: Praveen Vuppala <praveenv@xilinx.com> +// Author: Maruthi Srinivas Bayyavarapu <maruthis@xilinx.com> #include <linux/io.h> #include <linux/module.h> diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 96340f23f86d..bb8372833fc2 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -768,7 +768,7 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) * REG1: PLL binary search enable, soft mute enable. */ CM6206_REG1_PLLBIN_EN | - CM6206_REG1_SOFT_MUTE_EN | + CM6206_REG1_SOFT_MUTE_EN, /* * REG2: enable output drivers, * select front channels to the headphone output, @@ -1492,6 +1492,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x152a, 0x85de): /* SMSL D1 DAC */ case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ |