February 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation
by Adam Serbinski 12 Feb '20

12 Feb '20

This patch adds documentation of bindings required for PCM ports on AFE. Signed-off-by: Adam Serbinski <adam(a)serbinski.com> CC: Andy Gross <agross(a)kernel.org> CC: Mark Rutland <mark.rutland(a)arm.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Patrick Lai <plai(a)codeaurora.org> CC: Banajit Goswami <bgoswami(a)codeaurora.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.com> CC: alsa-devel(a)alsa-project.org CC: linux-arm-msm(a)vger.kernel.org CC: devicetree(a)vger.kernel.org CC: linux-kernel(a)vger.kernel.org --- .../devicetree/bindings/sound/qcom,q6afe.txt | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index d74888b9f1bb..6b1b17d31a2a 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -51,6 +51,24 @@ configuration of each dai. Must contain the following properties. Definition: Must be list of serial data lines used by this dai. should be one or more of the 0-3 sd lines. + - qcom,pcm-quantype + Usage: required for pcm interface + Value type: <u32> + Definition: PCM quantization type + 0 - ALAW, no padding + 1 - MULAW, no padding + 2 - Linear, no padding + 3 - ALAW, padding + 4 - MULAW, padding + 5 - Linear, padding + + - qcom,pcm-slot-mapping + Usage: required for pcm interface + Value type: <prop-encoded-array> + Definition: Slot mapping for audio channels. Array size is the number + of slots, minimum 1, maximum 4. The value is 0 for no mapping + to the slot, or the channel number from 1 to 32. + - qcom,tdm-sync-mode: Usage: required for tdm interface Value type: <prop-encoded-array> @@ -174,5 +192,29 @@ q6afe@4 { reg = <23>; qcom,sd-lines = <1>; }; + + pri-pcm-rx@105 { + reg = <105>; + qcom,pcm-quantype = <2>; + qcom,pcm-slot-mapping = <1>; + }; + + pri-pcm-tx@106 { + reg = <106>; + qcom,pcm-quantype = <2>; + qcom,pcm-slot-mapping = <1>; + }; + + quat-pcm-rx@111 { + reg = <111>; + qcom,pcm-quantype = <5>; + qcom,pcm-slot-mapping = <0 0 1>; + }; + + quat-pcm-tx@112 { + reg = <112>; + qcom,pcm-quantype = <5>; + qcom,pcm-slot-mapping = <0 0 1>; + }; }; }; -- 2.21.1

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[alsa-devel] [PATCH] ASoC: amd: Buffer Size instead of MAX Buffer
by Ravulapati Vishnu vardhan rao 12 Feb '20

12 Feb '20

Because of MAX BUFFER size in register,when user/app give small buffer size produces noise of old data in buffer. This patch rectifies this noise when using different buffer sizes less than MAX BUFFER. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 8 ++++++++ sound/soc/amd/raven/acp3x-pcm-dma.c | 7 +------ 2 files changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 31cd400..91a3881 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -170,6 +170,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, struct snd_soc_card *card; struct acp3x_platform_info *pinfo; u32 ret, val, period_bytes, reg_val, ier_val, water_val; + u32 buf_size, buf_reg; prtd = substream->private_data; rtd = substream->runtime->private_data; @@ -183,6 +184,8 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, } period_bytes = frames_to_bytes(substream->runtime, substream->runtime->period_size); + buf_size = frames_to_bytes(substream->runtime, + substream->runtime->buffer_size); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -196,6 +199,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, mmACP_BT_TX_INTR_WATERMARK_SIZE; reg_val = mmACP_BTTDM_ITER; ier_val = mmACP_BTTDM_IER; + buf_reg = mmACP_BT_TX_RINGBUFSIZE; break; case I2S_SP_INSTANCE: default: @@ -203,6 +207,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, mmACP_I2S_TX_INTR_WATERMARK_SIZE; reg_val = mmACP_I2STDM_ITER; ier_val = mmACP_I2STDM_IER; + buf_reg = mmACP_I2S_TX_RINGBUFSIZE; } } else { switch (rtd->i2s_instance) { @@ -211,6 +216,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, mmACP_BT_RX_INTR_WATERMARK_SIZE; reg_val = mmACP_BTTDM_IRER; ier_val = mmACP_BTTDM_IER; + buf_reg = mmACP_BT_RX_RINGBUFSIZE; break; case I2S_SP_INSTANCE: default: @@ -218,9 +224,11 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, mmACP_I2S_RX_INTR_WATERMARK_SIZE; reg_val = mmACP_I2STDM_IRER; ier_val = mmACP_I2STDM_IER; + buf_reg = mmACP_I2S_RX_RINGBUFSIZE; } } rv_writel(period_bytes, rtd->acp3x_base + water_val); + rv_writel(buf_size, rtd->acp3x_base + buf_reg); val = rv_readl(rtd->acp3x_base + reg_val); val = val | BIT(0); rv_writel(val, rtd->acp3x_base + reg_val); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index aecc3c0..d62c0d9 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -110,7 +110,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) { u16 page_idx; u32 low, high, val, acp_fifo_addr, reg_fifo_addr; - u32 reg_ringbuf_size, reg_dma_size, reg_fifo_size; + u32 reg_dma_size, reg_fifo_size; dma_addr_t addr; addr = rtd->dma_addr; @@ -157,7 +157,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) if (direction == SNDRV_PCM_STREAM_PLAYBACK) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - reg_ringbuf_size = mmACP_BT_TX_RINGBUFSIZE; reg_dma_size = mmACP_BT_TX_DMA_SIZE; acp_fifo_addr = ACP_SRAM_PTE_OFFSET + BT_PB_FIFO_ADDR_OFFSET; @@ -169,7 +168,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) case I2S_SP_INSTANCE: default: - reg_ringbuf_size = mmACP_I2S_TX_RINGBUFSIZE; reg_dma_size = mmACP_I2S_TX_DMA_SIZE; acp_fifo_addr = ACP_SRAM_PTE_OFFSET + SP_PB_FIFO_ADDR_OFFSET; @@ -181,7 +179,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: - reg_ringbuf_size = mmACP_BT_RX_RINGBUFSIZE; reg_dma_size = mmACP_BT_RX_DMA_SIZE; acp_fifo_addr = ACP_SRAM_PTE_OFFSET + BT_CAPT_FIFO_ADDR_OFFSET; @@ -193,7 +190,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) case I2S_SP_INSTANCE: default: - reg_ringbuf_size = mmACP_I2S_RX_RINGBUFSIZE; reg_dma_size = mmACP_I2S_RX_DMA_SIZE; acp_fifo_addr = ACP_SRAM_PTE_OFFSET + SP_CAPT_FIFO_ADDR_OFFSET; @@ -203,7 +199,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) rtd->acp3x_base + mmACP_I2S_RX_RINGBUFADDR); } } - rv_writel(MAX_BUFFER, rtd->acp3x_base + reg_ringbuf_size); rv_writel(DMA_SIZE, rtd->acp3x_base + reg_dma_size); rv_writel(acp_fifo_addr, rtd->acp3x_base + reg_fifo_addr); rv_writel(FIFO_SIZE, rtd->acp3x_base + reg_fifo_size); -- 2.7.4

3 3

[alsa-devel] [PATCH 24/28] sound: Call cpu_latency_qos_*() instead of pm_qos_*()
by Rafael J. Wysocki 12 Feb '20

12 Feb '20

From: "Rafael J. Wysocki" <rafael.j.wysocki(a)intel.com> Call cpu_latency_qos_add/update/remove_request() and cpu_latency_qos_request_active() instead of pm_qos_add/update/remove_request() and pm_qos_request_active(), respectively, because the latter are going to be dropped. No intentional functional impact. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- sound/core/pcm_native.c | 14 +++++++------- sound/soc/intel/atom/sst/sst.c | 5 ++--- sound/soc/intel/atom/sst/sst_loader.c | 4 ++-- sound/soc/ti/omap-dmic.c | 7 ++++--- sound/soc/ti/omap-mcbsp.c | 16 ++++++++-------- sound/soc/ti/omap-mcpdm.c | 16 ++++++++-------- 6 files changed, 31 insertions(+), 31 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 336406bcb59e..151bac1bbd0b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -748,11 +748,11 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); - if (pm_qos_request_active(&substream->latency_pm_qos_req)) - pm_qos_remove_request(&substream->latency_pm_qos_req); + if (cpu_latency_qos_request_active(&substream->latency_pm_qos_req)) + cpu_latency_qos_remove_request(&substream->latency_pm_qos_req); if ((usecs = period_to_usecs(runtime)) >= 0) - pm_qos_add_request(&substream->latency_pm_qos_req, - PM_QOS_CPU_DMA_LATENCY, usecs); + cpu_latency_qos_add_request(&substream->latency_pm_qos_req, + usecs); return 0; _error: /* hardware might be unusable from this time, @@ -821,7 +821,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) return -EBADFD; result = do_hw_free(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); - pm_qos_remove_request(&substream->latency_pm_qos_req); + cpu_latency_qos_remove_request(&substream->latency_pm_qos_req); return result; } @@ -2598,8 +2598,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) substream->ops->close(substream); substream->hw_opened = 0; } - if (pm_qos_request_active(&substream->latency_pm_qos_req)) - pm_qos_remove_request(&substream->latency_pm_qos_req); + if (cpu_latency_qos_request_active(&substream->latency_pm_qos_req)) + cpu_latency_qos_remove_request(&substream->latency_pm_qos_req); if (substream->pcm_release) { substream->pcm_release(substream); substream->pcm_release = NULL; diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index 68bcec5241f7..d6563985e008 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -325,8 +325,7 @@ int sst_context_init(struct intel_sst_drv *ctx) ret = -ENOMEM; goto do_free_mem; } - pm_qos_add_request(ctx->qos, PM_QOS_CPU_DMA_LATENCY, - PM_QOS_DEFAULT_VALUE); + cpu_latency_qos_add_request(ctx->qos, PM_QOS_DEFAULT_VALUE); dev_dbg(ctx->dev, "Requesting FW %s now...\n", ctx->firmware_name); ret = request_firmware_nowait(THIS_MODULE, true, ctx->firmware_name, @@ -364,7 +363,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sysfs_remove_group(&ctx->dev->kobj, &sst_fw_version_attr_group); flush_scheduled_work(); destroy_workqueue(ctx->post_msg_wq); - pm_qos_remove_request(ctx->qos); + cpu_latency_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); ctx->fw_sg_list.list_len = 0; diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index ce11c36848c4..9b0e3739c738 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -412,7 +412,7 @@ int sst_load_fw(struct intel_sst_drv *sst_drv_ctx) return -ENOMEM; /* Prevent C-states beyond C6 */ - pm_qos_update_request(sst_drv_ctx->qos, 0); + cpu_latency_qos_update_request(sst_drv_ctx->qos, 0); sst_drv_ctx->sst_state = SST_FW_LOADING; @@ -442,7 +442,7 @@ int sst_load_fw(struct intel_sst_drv *sst_drv_ctx) restore: /* Re-enable Deeper C-states beyond C6 */ - pm_qos_update_request(sst_drv_ctx->qos, PM_QOS_DEFAULT_VALUE); + cpu_latency_qos_update_request(sst_drv_ctx->qos, PM_QOS_DEFAULT_VALUE); sst_free_block(sst_drv_ctx, block); dev_dbg(sst_drv_ctx->dev, "fw load successful!!!\n"); diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c index 3f226be123d4..913579c43e9d 100644 --- a/sound/soc/ti/omap-dmic.c +++ b/sound/soc/ti/omap-dmic.c @@ -112,7 +112,7 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); - pm_qos_remove_request(&dmic->pm_qos_req); + cpu_latency_qos_remove_request(&dmic->pm_qos_req); if (!dai->active) dmic->active = 0; @@ -230,8 +230,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream, struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); u32 ctrl; - if (pm_qos_request_active(&dmic->pm_qos_req)) - pm_qos_update_request(&dmic->pm_qos_req, dmic->latency); + if (cpu_latency_qos_request_active(&dmic->pm_qos_req)) + cpu_latency_qos_update_request(&dmic->pm_qos_req, + dmic->latency); /* Configure uplink threshold */ omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold); diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 26b503bbdb5f..302d5c493c29 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -836,10 +836,10 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; if (mcbsp->latency[stream2]) - pm_qos_update_request(&mcbsp->pm_qos_req, - mcbsp->latency[stream2]); + cpu_latency_qos_update_request(&mcbsp->pm_qos_req, + mcbsp->latency[stream2]); else if (mcbsp->latency[stream1]) - pm_qos_remove_request(&mcbsp->pm_qos_req); + cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); mcbsp->latency[stream1] = 0; @@ -863,10 +863,10 @@ static int omap_mcbsp_dai_prepare(struct snd_pcm_substream *substream, if (!latency || mcbsp->latency[stream1] < latency) latency = mcbsp->latency[stream1]; - if (pm_qos_request_active(pm_qos_req)) - pm_qos_update_request(pm_qos_req, latency); + if (cpu_latency_qos_request_active(pm_qos_req)) + cpu_latency_qos_update_request(pm_qos_req, latency); else if (latency) - pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); + cpu_latency_qos_add_request(pm_qos_req, latency); return 0; } @@ -1434,8 +1434,8 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); - if (pm_qos_request_active(&mcbsp->pm_qos_req)) - pm_qos_remove_request(&mcbsp->pm_qos_req); + if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) + cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index a726cd7a8252..d7ac4df6f2d9 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -281,10 +281,10 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, } if (mcpdm->latency[stream2]) - pm_qos_update_request(&mcpdm->pm_qos_req, - mcpdm->latency[stream2]); + cpu_latency_qos_update_request(&mcpdm->pm_qos_req, + mcpdm->latency[stream2]); else if (mcpdm->latency[stream1]) - pm_qos_remove_request(&mcpdm->pm_qos_req); + cpu_latency_qos_remove_request(&mcpdm->pm_qos_req); mcpdm->latency[stream1] = 0; @@ -386,10 +386,10 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, if (!latency || mcpdm->latency[stream1] < latency) latency = mcpdm->latency[stream1]; - if (pm_qos_request_active(pm_qos_req)) - pm_qos_update_request(pm_qos_req, latency); + if (cpu_latency_qos_request_active(pm_qos_req)) + cpu_latency_qos_update_request(pm_qos_req, latency); else if (latency) - pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); + cpu_latency_qos_add_request(pm_qos_req, latency); if (!omap_mcpdm_active(mcpdm)) { omap_mcpdm_start(mcpdm); @@ -451,8 +451,8 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); - if (pm_qos_request_active(&mcpdm->pm_qos_req)) - pm_qos_remove_request(&mcpdm->pm_qos_req); + if (cpu_latency_qos_request_active(&mcpdm->pm_qos_req)) + cpu_latency_qos_remove_request(&mcpdm->pm_qos_req); return 0; } -- 2.16.4

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[alsa-devel] [PATCH v3 3/6] ASoC: qdsp6: q6afe-dai: add support to pcm port dais
by Adam Serbinski 12 Feb '20

12 Feb '20

This patch adds support of AFE DAI for PCM port. Signed-off-by: Adam Serbinski <adam(a)serbinski.com> CC: Andy Gross <agross(a)kernel.org> CC: Mark Rutland <mark.rutland(a)arm.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Patrick Lai <plai(a)codeaurora.org> CC: Banajit Goswami <bgoswami(a)codeaurora.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.com> CC: alsa-devel(a)alsa-project.org CC: linux-arm-msm(a)vger.kernel.org CC: devicetree(a)vger.kernel.org CC: linux-kernel(a)vger.kernel.org --- sound/soc/qcom/qdsp6/q6afe-dai.c | 210 ++++++++++++++++++++++++++++++- 1 file changed, 205 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index c1a7624eaf17..2deb8d6da462 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -63,6 +63,8 @@ struct q6afe_dai_priv_data { uint32_t invert_sync; uint32_t data_delay; uint32_t data_align; + uint32_t quantype; + uint32_t slot_number_mapping[AFE_MAX_PCM_SLOT_COUNT]; }; struct q6afe_dai_data { @@ -151,6 +153,52 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } +static int q6pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg; + int i = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + pcm->sample_rate = params_rate(params); + break; + default: + dev_err(dai->dev, "invalid sample rate = %u\n", + params_rate(params)); + return -EINVAL; + } + + pcm->quantype = dai_data->priv[dai->id].quantype; + + for (i = 0; i < AFE_MAX_PCM_SLOT_COUNT; i++) + pcm->slot_number_mapping[i] = + dai_data->priv[dai->id].slot_number_mapping[i]; + + return 0; +} + +static int q6pcm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + struct q6afe_pcm_cfg *pcm = &dai_data->port_config[dai->id].pcm_cfg; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + pcm->fmt = fmt; + break; + default: + dev_err(dai->dev, "unsupported format = %u\n", fmt); + return -EINVAL; + } + + return 0; +} + static int q6i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -358,6 +406,15 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, return rc; } break; + case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX: + rc = q6afe_pcm_port_prepare(dai_data->port[dai->id], + &dai_data->port_config[dai->id].pcm_cfg); + if (rc < 0) { + dev_err(dai->dev, "fail to prepare AFE port %x\n", + dai->id); + return rc; + } + break; case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7: q6afe_tdm_port_prepare(dai_data->port[dai->id], &dai_data->port_config[dai->id].tdm); @@ -429,16 +486,25 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, Q6AFE_LPASS_CLK_ROOT_DEFAULT, freq, dir); case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR: + case Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT ... Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR: + switch (dai->id) { + case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7: + return q6afe_port_set_sysclk(port, clk_id, + Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO, + Q6AFE_LPASS_CLK_ROOT_DEFAULT, + freq, dir); + break; + default: + return q6afe_port_set_sysclk(port, clk_id, + Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO, + Q6AFE_LPASS_CLK_ROOT_DEFAULT, + freq, dir); + } case Q6AFE_LPASS_CLK_ID_MCLK_1 ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1: return q6afe_port_set_sysclk(port, clk_id, Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO, Q6AFE_LPASS_CLK_ROOT_DEFAULT, freq, dir); - case Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT ... Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT: - return q6afe_port_set_sysclk(port, clk_id, - Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO, - Q6AFE_LPASS_CLK_ROOT_DEFAULT, - freq, dir); } return 0; @@ -468,6 +534,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"}, {"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"}, + {"Primary PCM Playback", NULL, "PRI_PCM_RX"}, + {"Secondary PCM Playback", NULL, "SEC_PCM_RX"}, + {"Tertiary PCM Playback", NULL, "TERT_PCM_RX"}, + {"Quaternary PCM Playback", NULL, "QUAT_PCM_RX"}, + {"Primary TDM0 Playback", NULL, "PRIMARY_TDM_RX_0"}, {"Primary TDM1 Playback", NULL, "PRIMARY_TDM_RX_1"}, {"Primary TDM2 Playback", NULL, "PRIMARY_TDM_RX_2"}, @@ -562,6 +633,11 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"PRI_MI2S_TX", NULL, "Primary MI2S Capture"}, {"SEC_MI2S_TX", NULL, "Secondary MI2S Capture"}, {"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"}, + + {"PRI_PCM_TX", NULL, "Primary PCM Capture"}, + {"SEC_PCM_TX", NULL, "Secondary PCM Capture"}, + {"TERT_PCM_TX", NULL, "Tertiary PCM Capture"}, + {"QUAT_PCM_TX", NULL, "Quaternary PCM Capture"}, }; static const struct snd_soc_dai_ops q6hdmi_ops = { @@ -578,6 +654,14 @@ static const struct snd_soc_dai_ops q6i2s_ops = { .set_sysclk = q6afe_mi2s_set_sysclk, }; +static const struct snd_soc_dai_ops q6pcm_ops = { + .prepare = q6afe_dai_prepare, + .hw_params = q6pcm_hw_params, + .set_fmt = q6pcm_set_fmt, + .shutdown = q6afe_dai_shutdown, + .set_sysclk = q6afe_mi2s_set_sysclk, +}; + static const struct snd_soc_dai_ops q6slim_ops = { .prepare = q6afe_dai_prepare, .hw_params = q6slim_hw_params, @@ -1012,6 +1096,94 @@ static struct snd_soc_dai_driver q6afe_dais[] = { .ops = &q6i2s_ops, .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Primary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .id = PRIMARY_PCM_RX, + .name = "PRI_PCM_RX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Primary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .id = PRIMARY_PCM_TX, + .name = "PRI_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Secondary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .name = "SEC_PCM_RX", + .id = SECONDARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Secondary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .id = SECONDARY_PCM_TX, + .name = "SEC_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Tertiary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .name = "TERT_PCM_RX", + .id = TERTIARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Tertiary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .id = TERTIARY_PCM_TX, + .name = "TERT_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .playback = { + .stream_name = "Quaternary PCM Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .name = "QUAT_PCM_RX", + .id = QUATERNARY_PCM_RX, + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, { + .capture = { + .stream_name = "Quaternary PCM Capture", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .id = QUATERNARY_PCM_TX, + .name = "QUAT_PCM_TX", + .ops = &q6pcm_ops, + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, }, Q6AFE_TDM_PB_DAI("Primary", 0, PRIMARY_TDM_RX_0), Q6AFE_TDM_PB_DAI("Primary", 1, PRIMARY_TDM_RX_1), @@ -1169,6 +1341,15 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("QUAT_PCM_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("QUAT_PCM_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("TERT_PCM_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("TERT_PCM_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("SEC_PCM_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("SEC_PCM_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("PRI_PCM_RX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("PRI_PCM_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL, @@ -1381,6 +1562,25 @@ static void of_q6afe_parse_dai_data(struct device *dev, for (i = 0; i < num_lines; i++) priv->sd_line_mask |= BIT(lines[i]); + break; + case PRIMARY_PCM_RX ... QUATERNARY_PCM_TX: + priv = &data->priv[id]; + ret = of_property_read_u32(node, "qcom,pcm-quantype", + &priv->quantype); + if (ret) { + dev_err(dev, "No quantype from DT\n"); + break; + } + + ret = of_property_read_variable_u32_array(node, + "qcom,pcm-slot-mapping", + priv->slot_number_mapping, + 1, AFE_MAX_PCM_SLOT_COUNT); + if (ret < 0) { + dev_err(dev, "No slot mapping from DT\n"); + break; + } + break; case PRIMARY_TDM_RX_0 ... QUINARY_TDM_TX_7: priv = &data->priv[id]; -- 2.21.1

2 2

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix silent output on MSI-GL73
by Takashi Iwai 12 Feb '20

12 Feb '20

MSI-GL73 laptop with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204159 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ee88adf57e7..6c8cb4ce517e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), -- 2.16.4

1 0

[alsa-devel] Add more codec supported Headset Button
by Kailang 12 Feb '20

12 Feb '20

Hi Takashi, Attach patch. Add supported Headset Button for ALC215/ALC285/ALC289. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ALSA: hda_codec: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 701a69d856f5..0fbca35509cd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -102,7 +102,7 @@ struct hda_conn_list { struct list_head list; int len; hda_nid_t nid; - hda_nid_t conns[0]; + hda_nid_t conns[]; }; /* look up the cached results */ -- 2.25.0

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e1ebc6d5f382..ab5632502206 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2708,7 +2708,7 @@ struct dsp_image_seg { u32 magic; u32 chip_addr; u32 count; - u32 data[0]; + u32 data[]; }; static const u32 g_magic_value = 0x4c46584d; -- 2.25.0

2 1

[alsa-devel] [PATCH] ALSA: usb-midi: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0ec77d0..f15d36731cbe 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -91,7 +91,7 @@ struct usb_ms_endpoint_descriptor { __u8 bDescriptorType; __u8 bDescriptorSubtype; __u8 bNumEmbMIDIJack; - __u8 baAssocJackID[0]; + __u8 baAssocJackID[]; } __attribute__ ((packed)); struct snd_usb_midi_in_endpoint; -- 2.25.0

2 1

[alsa-devel] [PATCH] ALSA: core: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 12 Feb '20

12 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/oss/rate.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 2fa9299a440d..e9f20fefb117 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -47,7 +47,7 @@ struct rate_priv { unsigned int pos; rate_f func; snd_pcm_sframes_t old_src_frames, old_dst_frames; - struct rate_channel channels[0]; + struct rate_channel channels[]; }; static void rate_init(struct snd_pcm_plugin *plugin) -- 2.25.0

2 1