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

[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

2 7

[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 01 Oct '19

01 Oct '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 2

[alsa-devel] [PATCH 1/5] ASoC: amd: Registering device endpoints using MFD framework
by Ravulapati Vishnu vardhan rao 30 Sep '19

30 Sep '19

Removed platform based endpoint registering. Now Registering DMA and multiple I2S BT playback endpoint devices automatically by using MFD framework. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Reviewed-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 54 ++++++++------- sound/soc/amd/raven/pci-acp3x.c | 148 +++++++++++++++++++++++++++++----------- 2 files changed, 139 insertions(+), 63 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd..3a27fc8 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,33 +7,41 @@ #include "chip_offset_byte.h" -#define ACP3x_PHY_BASE_ADDRESS 0x1240000 -#define ACP3x_I2S_MODE 0 -#define ACP3x_REG_START 0x1240000 -#define ACP3x_REG_END 0x1250200 -#define I2S_MODE 0x04 -#define BT_TX_THRESHOLD 26 -#define BT_RX_THRESHOLD 25 -#define ACP3x_POWER_ON 0x00 -#define ACP3x_POWER_ON_IN_PROGRESS 0x01 -#define ACP3x_POWER_OFF 0x02 -#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_DEVS 3 +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 +#define I2S_MODE 0x04 +#define I2S_RX_THRESHOLD 27 +#define I2S_TX_THRESHOLD 28 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP_ERR_INTR_MASK 29 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 #define ACP_SRAM_PTE_OFFSET 0x02050000 -#define PAGE_SIZE_4K_ENABLE 0x2 +#define PAGE_SIZE_4K_ENABLE 0x2 #define MEM_WINDOW_START 0x4000000 -#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 -#define CAPTURE_FIFO_ADDR_OFFSET 0x500 - -#define PLAYBACK_MIN_NUM_PERIODS 2 -#define PLAYBACK_MAX_NUM_PERIODS 8 -#define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 4096 -#define CAPTURE_MIN_NUM_PERIODS 2 -#define CAPTURE_MAX_NUM_PERIODS 8 -#define CAPTURE_MAX_PERIOD_SIZE 16384 -#define CAPTURE_MIN_PERIOD_SIZE 4096 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..c0b8fad 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -9,23 +9,49 @@ #include <linux/io.h> #include <linux/platform_device.h> #include <linux/interrupt.h> +#include <linux/mfd/core.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> #include "acp3x.h" +struct i2s_platform_data { + unsigned int cap; + int channel; + u32 snd_rates; +}; struct acp3x_dev_data { + struct device *parent; + struct mfd_cell *cell; + struct resource *res; void __iomem *acp3x_base; bool acp3x_audio_mode; - struct resource *res; struct platform_device *pdev; }; + +static struct device *get_mfd_cell_dev(const char *device_name, int r) +{ + char auto_dev_name[25]; + struct device *dev; + + snprintf(auto_dev_name, sizeof(auto_dev_name), + "%s.%d.auto", device_name, r); + dev = bus_find_device_by_name(&platform_bus_type, + NULL, auto_dev_name); + dev_info(dev, "device %s added\n", auto_dev_name); + return dev; +} + static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr, val; + uint64_t addr; + int val, i, r; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct device *dev; + struct i2s_platform_data *i2s_pdata; unsigned int irqflags; if (pci_enable_device(pci)) { @@ -63,47 +89,82 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - + adata->parent = &pci->dev; val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, - GFP_KERNEL); - if (!adata->res) { + sizeof(struct resource) * 4, + GFP_KERNEL); + adata->cell = devm_kzalloc(&pci->dev, + sizeof(struct mfd_cell) * 3, + GFP_KERNEL); + if (!adata->cell) { ret = -ENOMEM; goto unmap_mmio; } - adata->res[0].name = "acp3x_i2s_iomem"; - adata->res[0].flags = IORESOURCE_MEM; - adata->res[0].start = addr; - adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; - - adata->acp3x_audio_mode = ACP3x_I2S_MODE; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + i2s_pdata = devm_kzalloc(&pci->dev, + sizeof(struct i2s_platform_data) * 3, + GFP_KERNEL); + if (i2s_pdata == NULL) { + kfree(adata->res); + kfree(adata->cell); + return -ENOMEM; } + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + adata->res[0].start = addr; + adata->res[0].end = addr + + (ACP3x_REG_END - ACP3x_REG_START); + i2s_pdata[0].cap = 0; + i2s_pdata[0].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[1].name = "acp3x_i2s_sp_play_cap"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + i2s_pdata[1].cap = 0; + i2s_pdata[1].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[2].name = "acp3x_i2s_bt_play_cap"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + i2s_pdata[2].cap = 0; + i2s_pdata[2].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; + + adata->acp3x_audio_mode = ACP3x_I2S_MODE; + + adata->cell[0].name = "acp3x_rv_i2s_dma"; + adata->cell[0].num_resources = 4; + adata->cell[0].resources = &adata->res[0]; + adata->cell[0].platform_data = &irqflags; + adata->cell[0].pdata_size = sizeof(irqflags); + + adata->cell[1].name = "acp3x_i2s_playcap"; + adata->cell[1].num_resources = 1; + adata->cell[1].resources = &adata->res[1]; + adata->cell[1].platform_data = &i2s_pdata[0]; + adata->cell[1].pdata_size = + sizeof(struct i2s_platform_data); + + adata->cell[2].name = "acp3x_i2s_playcap"; + adata->cell[2].num_resources = 1; + adata->cell[2].resources = &adata->res[2]; + adata->cell[2].platform_data = &i2s_pdata[1]; + adata->cell[2].pdata_size = + sizeof(struct i2s_platform_data); + r = mfd_add_hotplug_devices(adata->parent, adata->cell, 3); + for (i = 0; i < 3 ; i++) + dev = get_mfd_cell_dev(adata->cell[i].name, i); break; + default: dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; @@ -112,23 +173,29 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); + /*ignore device status and return driver probe error*/ + return -ENODEV; release_regions: pci_release_regions(pci); + /*ignore device status and return driver probe error*/ + return -ENODEV; disable_pci: pci_disable_device(pci); - - return ret; + /*ignore device status and return driver probe error*/ + return -ENODEV; } static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); - - platform_device_unregister(adata->pdev); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); @@ -151,6 +218,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.7.4

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Re: [alsa-devel] [PATCH v2] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Pierre-Louis Bossart 28 Sep '19

28 Sep '19

On 9/26/19 9:55 PM, Navid Emamdoost wrote: > On Wed, Sep 25, 2019 at 12:05:28PM -0500, Pierre-Louis Bossart wrote: >> On 9/25/19 11:19 AM, Navid Emamdoost wrote: >>> In snd_skl_parse_uuids if allocation for module->instance_id fails, the >>> allocated memory for module shoulde be released. I changes the >>> allocation for module to use devm_kzalloc to be resource_managed >>> allocation and avoid the release in error path. >> >> if you use devm_, don't you need to fix the error path as well then, I see a >> kfree(uuid) in skl_freeup_uuid_list(). >> >> I am not very familiar with this code but the error seems to be that the >> list_add_tail() is called after the module->instance_id is allocated, so >> there is a risk that the module allocated earlier is not freed (since it's >> not yet added to the list). Freeing the module as done in patch 1 works, >> using devm_ without fixing the error path does not seem correct to me. >> > Thanks for the feedback, then it's your call if you can accept patch 1 as > fix. Cezary, it's really your call. > Navid. >>> >>> Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> >>> --- >>> Changes in v2: >>> - Changed the allocation for module from kzalloc to devm_kzalloc >>> --- >>> sound/soc/intel/skylake/skl-sst-utils.c | 2 +- >>> 1 file changed, 1 insertion(+), 1 deletion(-) >>> >>> diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c >>> index d43cbf4a71ef..ac37f04b0eea 100644 >>> --- a/sound/soc/intel/skylake/skl-sst-utils.c >>> +++ b/sound/soc/intel/skylake/skl-sst-utils.c >>> @@ -284,7 +284,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, >>> */ >>> for (i = 0; i < num_entry; i++, mod_entry++) { >>> - module = kzalloc(sizeof(*module), GFP_KERNEL); >>> + module = devm_kzalloc(ctx->dev, sizeof(*module), GFP_KERNEL); >>> if (!module) { >>> ret = -ENOMEM; >>> goto free_uuid_list; >>> >>

4 5

[alsa-devel] [PATCH v5] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 27 Sep '19

27 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v5: - Removed '#clock-cells' property from required properties - Added deprecated property to 'clock-output-names' and removed corresponding comment. Best regards, Maciej Falkowski --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 136 ++++++++++++++++++ 2 files changed, 136 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..b9502b82fa9d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,136 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for i2s0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for i2s1 and i2s2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clk. i2s0 has internal mux to select the source + of root clk and i2s1 and i2s2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for i2s0 + - const: i2s_cdclk0 + - items: # for i2s1 + - const: i2s_cdclk1 + - items: # for i2s2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

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[alsa-devel] [PATCH] ASoC: Skylake SST driver - blacklist the PCI device IDs for the auto probe
by Jaroslav Kysela 27 Sep '19

27 Sep '19

There are basically three drivers for the PCI devices for the recent Intel hardware with the build-in DSPs. The legacy HDA driver has dmic_detect module option for the auto detection of the platforms with the digital microphone. Because the SOF driver is preferred, just skip PCI probe in the Skylake SST driver when the PCI device ID clashes by default. The user can override the auto behaviour with the pci_binding module parameter. Boot log from Lenovo Carbon X1 (7th gen) with the default settings: snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, aborting probe snd_soc_skl 0000:00:1f.3: SOF driver is preferred on this platform, aborting probe sof-audio-pci 0000:00:1f.3: warning: No matching ASoC machine driver found sof-audio-pci 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040380 .... Perhaps, it may be more wise to create one shared module and all three drivers should call the driver detection routine(s) from one place. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/skylake/skl.c | 37 +++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..cace55ca8d55 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -38,6 +38,39 @@ static int skl_pci_binding; module_param_named(pci_binding, skl_pci_binding, int, 0444); MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); +/* + * + */ +static int skl_sof_support(struct pci_dev *pci) +{ + /* the SOF driver has same PCI IDs */ + if (pci->vendor == 0x8086) { + switch (pci->device) { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + case 0x02c8: /* CML-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + case 0x06c8: /* CML-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + case 0x3198: /* GLK */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + case 0x5a98: /* BXT-P */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + case 0x9dc8: /* CNL */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + case 0xa348: /* CFL */ +#endif + case 0x0000: /* a dummy value when no SOF driver enabled */ + return 1; + } + } + return 0; +} + /* * initialize the PCI registers */ @@ -1002,6 +1035,10 @@ static int skl_probe(struct pci_dev *pci, dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); return -ENODEV; } + if (skl_sof_support(pci)) { + dev_info(&pci->dev, "SOF driver is preferred on this platform, aborting probe\n"); + return -ENODEV; + } dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: -- 2.20.1

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[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up an indentation issue
by Colin King 27 Sep '19

27 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> There is a return statement that is indented too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

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[alsa-devel] sof-skl_hda: remove volume setting in enable/disable sequences
by GitHub pull_request - opened 27 Sep '19

27 Sep '19

alsa-project/alsa-ucm-conf pull request #2 was opened from juimonen: If the volume setting is defined in enable sequence you will always get that exact volume when switching outputs in some user space software like Pulseaudio. Pulseaudio will itself keep track of the volume previously set. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/2 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/2.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[alsa-devel] Playback volumes
by GitHub pull_request - opened 27 Sep '19

27 Sep '19

alsa-project/alsa-ucm-conf pull request #1 was opened from juimonen: Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/1 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[alsa-devel] [PATCH] ASoC: bdw-rt5677: channel constraint support
by Brent Lu 27 Sep '19

27 Sep '19

BDW boards using this machine driver supports only stereo capture and playback. Implement a constraint to enforce it. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..a312b55 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -22,6 +22,8 @@ #include "../../codecs/rt5677.h" +#define DUAL_CHANNEL 2 + struct bdw_rt5677_priv { struct gpio_desc *gpio_hp_en; struct snd_soc_component *component; @@ -245,6 +247,36 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5677_fe_ops = { + .startup = bdw_fe_startup, +}; + /* broadwell digital audio interface glue - connects codec <--> CPU */ SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); @@ -273,6 +305,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .ops = &bdw_rt5677_fe_ops, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

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