December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: cs42l51: add dac mux widget in codec routes
by Olivier Moysan 09 Dec '19

09 Dec '19

Add "DAC mux" DAPM widget in CS42l51 audio codec routes, to support DAC mux control and to remove error trace "DAC Mux has no paths" at widget creation. Note: ADC path of DAC mux is not routed in this patch. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/codecs/cs42l51.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 55408c8fcb4e..e47758e4fb36 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -214,12 +214,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture", CS42L51_POWER_CTL1, 2, 1, cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), - SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback", - CS42L51_POWER_CTL1, 5, 1, - cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), - SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback", - CS42L51_POWER_CTL1, 6, 1, - cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Left DAC", NULL, CS42L51_POWER_CTL1, 5, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Right DAC", NULL, CS42L51_POWER_CTL1, 6, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), /* analog/mic */ SND_SOC_DAPM_INPUT("AIN1L"), @@ -255,6 +253,12 @@ static const struct snd_soc_dapm_route cs42l51_routes[] = { {"HPL", NULL, "Left DAC"}, {"HPR", NULL, "Right DAC"}, + {"Right DAC", NULL, "DAC Mux"}, + {"Left DAC", NULL, "DAC Mux"}, + + {"DAC Mux", "Direct PCM", "Playback"}, + {"DAC Mux", "DSP PCM", "Playback"}, + {"Left ADC", NULL, "Left PGA"}, {"Right ADC", NULL, "Right PGA"}, -- 2.17.1

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[alsa-devel] [PATCH] ASoC: ti: davinci-mcasp: Improve the sysclk selection
by Peter Ujfalusi 09 Dec '19

09 Dec '19

When McASP is master the bclk can be generated from two main source: AUXCLK: functional clock for McASP or AHCLK: from external source or internal mux in dra7x family With this patch it is possible to select between the two source. The patch is not breaking existing machine drivers since historically the clk_id was ignored and left as 0 in all cases. When output clock is configured - which can be only the AHCLK, we select the AUXCLK as source for the internal HCLK. In this case the HCLK rate is the same as the output clock. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 35 ++++++++++++++++++++++++++++------- sound/soc/ti/davinci-mcasp.h | 4 ++++ 2 files changed, 32 insertions(+), 7 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 8e5371801d88..e1e937eb1dc1 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -664,18 +664,39 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); pm_runtime_get_sync(mcasp->dev); - if (dir == SND_SOC_CLOCK_OUT) { + + if (dir == SND_SOC_CLOCK_IN) { + switch (clk_id) { + case MCASP_CLK_HCLK_AHCLK: + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + clear_bit(PIN_BIT_AHCLKX, &mcasp->pdir); + break; + case MCASP_CLK_HCLK_AUXCLK: + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + set_bit(PIN_BIT_AHCLKX, &mcasp->pdir); + break; + default: + dev_err(mcasp->dev, "Invalid clk id: %d\n", clk_id); + goto out; + } + } else { + /* Select AUXCLK as HCLK */ mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); set_bit(PIN_BIT_AHCLKX, &mcasp->pdir); - } else { - mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); - mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); - clear_bit(PIN_BIT_AHCLKX, &mcasp->pdir); } - + /* + * When AHCLK X/R is selected to be output it means that the HCLK is + * the same clock - coming via AUXCLK. + */ mcasp->sysclk_freq = freq; - +out: pm_runtime_put(mcasp->dev); return 0; } diff --git a/sound/soc/ti/davinci-mcasp.h b/sound/soc/ti/davinci-mcasp.h index bc705d6ca48b..5de2b8a31061 100644 --- a/sound/soc/ti/davinci-mcasp.h +++ b/sound/soc/ti/davinci-mcasp.h @@ -295,6 +295,10 @@ #define NUMEVT(x) (((x) & 0xFF) << 8) #define NUMDMA_MASK (0xFF) +/* Source of High-frequency transmit/receive clock */ +#define MCASP_CLK_HCLK_AHCLK 0 /* AHCLKX/R */ +#define MCASP_CLK_HCLK_AUXCLK 1 /* Internal functional clock */ + /* clock divider IDs */ #define MCASP_CLKDIV_AUXCLK 0 /* HCLK divider from AUXCLK */ #define MCASP_CLKDIV_BCLK 1 /* BCLK divider from HCLK */ -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: SOF: Intel: hda: solve MSI issues by merging ipc and stream irq handlers
by Pierre-Louis Bossart 09 Dec '19

09 Dec '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> The existing code uses two handlers for a shared edge-based MSI interrupts. In corner cases, interrupts are lost, leading to IPC timeouts. Those timeouts do not appear in legacy mode. This patch merges the two handlers and threads into a single one, and simplifies the mask/unmask operations by using a single top-level mask (Global Interrupt Enable). The handler only checks for interrupt sources using the Global Interrupt Status (GIS) field, and all the actual work happens in the thread. This also enables us to remove the use of spin locks. Stream events are prioritized over IPC ones. This patch was tested with HDaudio and SoundWire platforms, and all known IPC timeout issues are solved in MSI mode. The SoundWire-specific patches will be provided in follow-up patches, where the SoundWire interrupts are handled in the same thread as IPC and stream interrupts. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/apl.c | 1 - sound/soc/sof/intel/cnl.c | 5 --- sound/soc/sof/intel/hda-ipc.c | 23 +++-------- sound/soc/sof/intel/hda-stream.c | 20 ++++----- sound/soc/sof/intel/hda.c | 69 ++++++++++++++++++++++---------- sound/soc/sof/intel/hda.h | 11 ++--- 6 files changed, 70 insertions(+), 59 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 9e4d859b4dba..aeed1422900b 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -42,7 +42,6 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .block_write = sof_block_write, /* doorbell */ - .irq_handler = hda_dsp_ipc_irq_handler, .irq_thread = hda_dsp_ipc_irq_thread, /* ipc */ diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 3541063ba84c..6b44f6d02082 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -107,10 +107,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) "nothing to do in IPC IRQ thread\n"); } - /* re-enable IPC interrupt */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); - return IRQ_HANDLED; } @@ -232,7 +228,6 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .block_write = sof_block_write, /* doorbell */ - .irq_handler = hda_dsp_ipc_irq_handler, .irq_thread = cnl_ipc_irq_thread, /* ipc */ diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 0fd2153c1769..1837f66e361f 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -230,22 +230,15 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) "nothing to do in IPC IRQ thread\n"); } - /* re-enable IPC interrupt */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); - return IRQ_HANDLED; } -/* is this IRQ for ADSP ? - we only care about IPC here */ -irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) +/* Check if an IPC IRQ occurred */ +bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev) { - struct snd_sof_dev *sdev = context; - int ret = IRQ_NONE; + bool ret = false; u32 irq_status; - spin_lock(&sdev->hw_lock); - /* store status */ irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS); dev_vdbg(sdev->dev, "irq handler: irq_status:0x%x\n", irq_status); @@ -255,16 +248,10 @@ irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) goto out; /* IPC message ? */ - if (irq_status & HDA_DSP_ADSPIS_IPC) { - /* disable IPC interrupt */ - snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, - HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, 0); - ret = IRQ_WAKE_THREAD; - } + if (irq_status & HDA_DSP_ADSPIS_IPC) + ret = true; out: - spin_unlock(&sdev->hw_lock); return ret; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 92643a801f4f..c0ab9bb2a797 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -550,22 +550,23 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, return 0; } -irqreturn_t hda_dsp_stream_interrupt(int irq, void *context) +bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) { - struct hdac_bus *bus = context; - int ret = IRQ_WAKE_THREAD; + struct hdac_bus *bus = sof_to_bus(sdev); + bool ret = false; u32 status; - spin_lock(&bus->reg_lock); + /* The function can be called at irq thread, so use spin_lock_irq */ + spin_lock_irq(&bus->reg_lock); status = snd_hdac_chip_readl(bus, INTSTS); dev_vdbg(bus->dev, "stream irq, INTSTS status: 0x%x\n", status); - /* Register inaccessible, ignore it.*/ - if (status == 0xffffffff) - ret = IRQ_NONE; + /* if Register inaccessible, ignore it.*/ + if (status != 0xffffffff) + ret = true; - spin_unlock(&bus->reg_lock); + spin_unlock_irq(&bus->reg_lock); return ret; } @@ -603,7 +604,8 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) { - struct hdac_bus *bus = context; + struct snd_sof_dev *sdev = context; + struct hdac_bus *bus = sof_to_bus(sdev); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) u32 rirb_status; #endif diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 84279f43576c..8d27846d9048 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -402,6 +402,49 @@ static const struct sof_intel_dsp_desc return chip_info; } +static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + + /* + * Get global interrupt status. It includes all hardware interrupt + * sources in the Intel HD Audio controller. + */ + if (snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS) & + SOF_HDA_INTSTS_GIS) { + + /* disable GIE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + SOF_HDA_INT_GLOBAL_EN, + 0); + + return IRQ_WAKE_THREAD; + } + + return IRQ_NONE; +} + +static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + + /* deal with streams and controller first */ + if (hda_dsp_check_stream_irq(sdev)) + hda_dsp_stream_threaded_handler(irq, sdev); + + if (hda_dsp_check_ipc_irq(sdev)) + sof_ops(sdev)->irq_thread(irq, sdev); + + /* enable GIE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + SOF_HDA_INT_GLOBAL_EN, + SOF_HDA_INT_GLOBAL_EN); + + return IRQ_HANDLED; +} + int hda_dsp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); @@ -506,9 +549,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) */ if (hda_use_msi && pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI) > 0) { dev_info(sdev->dev, "use msi interrupt mode\n"); - hdev->irq = pci_irq_vector(pci, 0); - /* ipc irq number is the same of hda irq */ - sdev->ipc_irq = hdev->irq; + sdev->ipc_irq = pci_irq_vector(pci, 0); /* initialised to "false" by kzalloc() */ sdev->msi_enabled = true; } @@ -519,28 +560,17 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) * in IO-APIC mode, hda->irq and ipc_irq are using the same * irq number of pci->irq */ - hdev->irq = pci->irq; sdev->ipc_irq = pci->irq; } - dev_dbg(sdev->dev, "using HDA IRQ %d\n", hdev->irq); - ret = request_threaded_irq(hdev->irq, hda_dsp_stream_interrupt, - hda_dsp_stream_threaded_handler, - IRQF_SHARED, "AudioHDA", bus); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to register HDA IRQ %d\n", - hdev->irq); - goto free_irq_vector; - } - dev_dbg(sdev->dev, "using IPC IRQ %d\n", sdev->ipc_irq); - ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_ipc_irq_handler, - sof_ops(sdev)->irq_thread, IRQF_SHARED, - "AudioDSP", sdev); + ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_interrupt_handler, + hda_dsp_interrupt_thread, + IRQF_SHARED, "AudioDSP", sdev); if (ret < 0) { dev_err(sdev->dev, "error: failed to register IPC IRQ %d\n", sdev->ipc_irq); - goto free_hda_irq; + goto free_irq_vector; } pci_set_master(pci); @@ -571,8 +601,6 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) free_ipc_irq: free_irq(sdev->ipc_irq, sdev); -free_hda_irq: - free_irq(hdev->irq, bus); free_irq_vector: if (sdev->msi_enabled) pci_free_irq_vectors(pci); @@ -618,7 +646,6 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) SOF_HDA_PPCTL_GPROCEN, 0); free_irq(sdev->ipc_irq, sdev); - free_irq(hda->irq, bus); if (sdev->msi_enabled) pci_free_irq_vectors(pci); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 97c3fcb23d15..01529c7058f3 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -43,11 +43,14 @@ /* SOF_HDA_GCTL register bist */ #define SOF_HDA_GCTL_RESET BIT(0) -/* SOF_HDA_INCTL and SOF_HDA_INTSTS regs */ +/* SOF_HDA_INCTL regs */ #define SOF_HDA_INT_GLOBAL_EN BIT(31) #define SOF_HDA_INT_CTRL_EN BIT(30) #define SOF_HDA_INT_ALL_STREAM 0xff +/* SOF_HDA_INTSTS regs */ +#define SOF_HDA_INTSTS_GIS BIT(31) + #define SOF_HDA_MAX_CAPS 10 #define SOF_HDA_CAP_ID_OFF 16 #define SOF_HDA_CAP_ID_MASK GENMASK(SOF_HDA_CAP_ID_OFF + 11,\ @@ -406,8 +409,6 @@ struct sof_intel_hda_dev { /* the maximum number of streams (playback + capture) supported */ u32 stream_max; - int irq; - /* PM related */ bool l1_support_changed;/* during suspend, is L1SEN changed or not */ @@ -511,11 +512,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params); int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream, int cmd); -irqreturn_t hda_dsp_stream_interrupt(int irq, void *context); irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context); int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct hdac_stream *stream); +bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); +bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction); @@ -540,7 +542,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev); int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev); int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); -irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context); irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context); int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: fix soc-core.c kernel-doc warning
by Randy Dunlap 09 Dec '19

09 Dec '19

From: Randy Dunlap <rdunlap(a)infradead.org> Fix a kernel-doc warning in soc-core.c by adding notation for @legacy_dai_naming. ../sound/soc/soc-core.c:2509: warning: Function parameter or member 'legacy_dai_naming' not described in 'snd_soc_register_dai' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) --- linux-next-20191209.orig/sound/soc/soc-core.c +++ linux-next-20191209/sound/soc/soc-core.c @@ -2508,6 +2508,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai * * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAI + * @legacy_dai_naming: if %true, use legacy single-name format; + * if %false, use multiple-name format; * * 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

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[alsa-devel] [PATCH v2] ASoC: hdmi-codec: re-introduce mutex locking again
by Jerome Brunet 09 Dec '19

09 Dec '19

The dai codec needs to ensure that on one dai is used at any time. This is currently protected by bit atomic operation. With this change, it done with a mutex instead. This change is not about functionality or efficiency. It is done with the hope that it help maintainability in the future. Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/codecs/hdmi-codec.c | 32 +++++++++++++++++++------------- 1 file changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index f8b5b960e597..543363102d03 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -274,7 +274,8 @@ struct hdmi_codec_priv { uint8_t eld[MAX_ELD_BYTES]; struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; - unsigned long busy; + struct mutex lock; + bool busy; struct snd_soc_jack *jack; unsigned int jack_status; }; @@ -390,9 +391,10 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); int ret = 0; - ret = test_and_set_bit(0, &hcp->busy); - if (ret) { + mutex_lock(&hcp->lock); + if (hcp->busy) { dev_err(dai->dev, "Only one simultaneous stream supported!\n"); + mutex_unlock(&hcp->lock); return -EINVAL; } @@ -405,21 +407,21 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, if (hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); + if (ret) + goto err; + + ret = snd_pcm_hw_constraint_eld(substream->runtime, hcp->eld); + if (ret) + goto err; - if (!ret) { - ret = snd_pcm_hw_constraint_eld(substream->runtime, - hcp->eld); - if (ret) - goto err; - } /* Select chmap supported */ hdmi_codec_eld_chmap(hcp); } - return 0; + + hcp->busy = true; err: - /* Release the exclusive lock on error */ - clear_bit(0, &hcp->busy); + mutex_unlock(&hcp->lock); return ret; } @@ -431,7 +433,9 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data); - clear_bit(0, &hcp->busy); + mutex_lock(&hcp->lock); + hcp->busy = false; + mutex_unlock(&hcp->lock); } static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, @@ -811,6 +815,8 @@ static int hdmi_codec_probe(struct platform_device *pdev) return -ENOMEM; hcp->hcd = *hcd; + mutex_init(&hcp->lock); + daidrv = devm_kcalloc(dev, dai_count, sizeof(*daidrv), GFP_KERNEL); if (!daidrv) return -ENOMEM; -- 2.23.0

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[alsa-devel] [PATCH v2 00/13] ASoC: SOF: initial cleanup for DT and multi-client support
by Pierre-Louis Bossart 09 Dec '19

09 Dec '19

This series does not add new functionality but introduces a better code partition to support machine drivers in Device Tree, as well as a 'componentization' of the PCM/topology parts needed for multi-client support. This cleanup was merged in the SOF tree and tested extensively with no regressions identified. Thanks Ranjani and Daniel for combining your proposals in a single change. Changes since v1: Rebased on top of Jaroslav's HDaudio changes (added in this series as discussed with Mark) Daniel Baluta (2): ASoC: SOF: Make creation of machine device from SOF core optional ASoC: SOF: Remove unused drv_name in sof_pdata Jaroslav Kysela (3): ASoC: intel/skl/hda - export number of digital microphones via control components ASoC: Intel - use control components to describe card config ASoC: Intel - do not describe I/O configuration in the long card name Ranjani Sridharan (8): ASoC: SOF: topology: remove snd_sof_init_topology() ASoC: SOF: core: modify the signature for snd_sof_create_page_table ASoC: SOF: core: move check for runtime callbacks to core ASoC: SOF: Introduce default_fw_filename member in sof_dev_desc ASoC: SOF: partition audio-related parts from SOF core ASoC: SOF: intel: hda: Modify signature for hda_codec_probe_bus() ASoC: SOF: remove nocodec_fw_filename ASoC: SOF: nocodec: Amend arguments for sof_nocodec_setup() include/sound/sof.h | 8 +- sound/soc/intel/boards/Kconfig | 13 + sound/soc/intel/boards/bytcht_es8316.c | 13 +- sound/soc/intel/boards/bytcr_rt5640.c | 10 + sound/soc/intel/boards/bytcr_rt5651.c | 22 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 8 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/control.c | 55 +-- sound/soc/sof/core.c | 244 +--------- sound/soc/sof/intel/apl.c | 7 + sound/soc/sof/intel/bdw.c | 33 ++ sound/soc/sof/intel/byt.c | 45 ++ sound/soc/sof/intel/cnl.c | 7 + sound/soc/sof/intel/hda-codec.c | 16 +- sound/soc/sof/intel/hda-dai.c | 1 + sound/soc/sof/intel/hda-pcm.c | 4 +- sound/soc/sof/intel/hda-stream.c | 1 + sound/soc/sof/intel/hda.c | 228 ++++++---- sound/soc/sof/intel/hda.h | 8 +- sound/soc/sof/ipc.c | 16 +- sound/soc/sof/nocodec.c | 12 - sound/soc/sof/ops.h | 34 ++ sound/soc/sof/pcm.c | 131 +++--- sound/soc/sof/pm.c | 219 +-------- sound/soc/sof/sof-acpi-dev.c | 39 +- sound/soc/sof/sof-audio.c | 445 +++++++++++++++++++ sound/soc/sof/sof-audio.h | 211 +++++++++ sound/soc/sof/sof-of-dev.c | 24 +- sound/soc/sof/sof-pci-dev.c | 47 +- sound/soc/sof/sof-priv.h | 202 +-------- sound/soc/sof/topology.c | 374 ++++++++-------- sound/soc/sof/trace.c | 4 +- sound/soc/sof/utils.c | 60 +++ 33 files changed, 1380 insertions(+), 1163 deletions(-) create mode 100644 sound/soc/sof/sof-audio.c create mode 100644 sound/soc/sof/sof-audio.h -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda/ca0132 - add missed snd_hda_gen_parse_auto_config
by Chuhong Yuan 09 Dec '19

09 Dec '19

It is found that all usages of snd_hda_parse_pin_defcfg() are followed with snd_hda_gen_parse_auto_config() except here. This should be a miss and needs to add the missed call. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/pci/hda/patch_ca0132.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index b7a1abb3e231..ea1187a13241 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -24,6 +24,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_generic.h" #include "ca0132_regs.h" @@ -8844,6 +8845,10 @@ static int patch_ca0132(struct hda_codec *codec) if (err < 0) goto error; + err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); + if (err < 0) + goto error; + return 0; error: -- 2.24.0

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[alsa-devel] [PATCH] ASoC: tas2552: add missed regulator_bulk_disable in remove
by Chuhong Yuan 09 Dec '19

09 Dec '19

The driver forgets to call regulator_bulk_disable() in remove like that in probe failure. Add the missed call to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/tas2552.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 56671f21cfe5..0e19ec76aae0 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -616,6 +616,9 @@ static void tas2552_component_remove(struct snd_soc_component *component) pm_runtime_put(component->dev); gpiod_set_value(tas2552->enable_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); }; #ifdef CONFIG_PM -- 2.24.0

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[alsa-devel] [PATCH] ASoC: cs35l32: add missed regulator_bulk_disable in remove
by Chuhong Yuan 09 Dec '19

09 Dec '19

The driver forgets to call regulator_bulk_disable() in remove like that in probe failure. Add the missed call to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/cs35l32.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 3a644a35c464..be8159bd8ec8 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -501,6 +501,8 @@ static int cs35l32_i2c_remove(struct i2c_client *i2c_client) /* Hold down reset */ gpiod_set_value_cansleep(cs35l32->reset_gpio, 0); + regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), + cs35l32->supplies); return 0; } -- 2.24.0

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[alsa-devel] [PATCH] firewire-motu: fix double unlocked 'motu->mutex'
by Takashi Sakamoto 09 Dec '19

09 Dec '19

Mutex is doubly unlocked in some error path of pcm.open. This commit fixes ALSA firewire-motu driver in Linux kernel v5.5. Reported-by: kbuild test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 3fd80b200388 ("ALSA: firewire-motu: use the same size of period for PCM substream in AMDTP streams") Fixes: 0f5482e7875b ("ALSA: firewire-motu: share PCM buffer size for both direction") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-pcm.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 349b4d09e84f..005970931030 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -177,18 +177,14 @@ static int pcm_open(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, frames_per_period, frames_per_period); - if (err < 0) { - mutex_unlock(&motu->mutex); + if (err < 0) goto err_locked; - } err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, frames_per_buffer, frames_per_buffer); - if (err < 0) { - mutex_unlock(&motu->mutex); + if (err < 0) goto err_locked; - } } } -- 2.20.1

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