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

[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

4 4

[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

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[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

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[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

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[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

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[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

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[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

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[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id
by Benjamin Poirier 12 Feb '20

12 Feb '20

fixes the pci subsystem device ids for the "Thinkpad X1 Carbon 7th" and "Thinkpad X1 Yoga 7th" quirks. My machine reports the following: dmidecode -t system Manufacturer: LENOVO Product Name: 20QDCTO1WW Version: ThinkPad X1 Carbon 7th lspci -s 1f.3 -vnn 00:1f.3 Audio device [0403]: Intel Corporation Cannon Point-LP High Definition Audio Controller [8086:9dc8] (rev 11) (prog-if 80) Subsystem: Lenovo Cannon Point-LP High Definition Audio Controller [17aa:2292] /proc/asound/card0/codec#0 Subsystem Id: 0x17aa2293 Notice the different subsystem device ids between pci info and codec info. commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") added a quirk meant for the X1 Carbon but used device id 0x2293. Note that this does not match the PCI SSID but it matches the codec SSID. commit 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") added a quirk meant for the X1 Yoga but used subdevice id 0x2292, the PCI SSID used on the X1 Carbon. Given that in snd_hdac_device_init() quirks are first matched by PCI SSID and then, if there is no match, by codec SSID, the net result is that the quirk labelled "Thinkpad X1 Yoga 7th" now gets applied on the X1 Carbon. Example from my machine (an X1 Carbon, not Yoga): [ 15.817637] snd_hda_codec_realtek hdaudioC0D0: ALC285: picked fixup Thinkpad X1 Yoga 7th (PCI SSID) Therefore, fix the subdevice id for the "ThinkPad X1 Carbon 7th" quirk. Note that looking through the lspci outputs collected at https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPad all X1 Carbon there have an Audio device with PCI SSID 0x2292, which matches with the output from my machine. This leaves the question of what to do with the quirk labelled "Thinkpad X1 Yoga 7th". >From email discussions, it seems that the author of commit 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") did not have a device to test the changes. I don't have an X1 Yoga either and I did not find a sample lspci listing for it online. Therefore, the best course of action seems to be to remove that quirk. In the best case, the quirk for the X1 Carbon will match the X1 Yoga (via PCI SSID or codec SSID). In the worst case, it will not and someone who actually has such a machine should come forth with concrete data about subsystem ids and needed quirks. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Fixes: 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen") Link: https://lore.kernel.org/alsa-devel/20200210025249.GA2700@f3/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4770fb3f51fb..05d44df2008e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7268,8 +7268,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), - SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_SPEAKER2_TO_DAC1), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.25.0

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[alsa-devel] MIDI 2.0 Universal MIDI Packet format (UMP)
by GitHub issues - edited 12 Feb '20

12 Feb '20

alsa-project/alsa-lib issue #24 was edited from symdeb: Since MIDI 2.0 only support USB and not over DIN (for now), will MIDI 2.0 need a new or updated OS class driver for the 32/64 bit universal packet interface and device descriptor for devices ? Page 16 of the USB MIDI class driver spec defines support for 1,2 or 3 bytes transfers only, How will this have to be implemented by MIDI USB device manufacturers and well as on the Linux OSes side ? How will this impact Linux ALSA ? Would the first byte of the message with b7=0 not confuse the driver of running status ? Link universal packet format https://www.midi.org/articles-old/details-about-midi-2-0-midi-ci-profiles-a…) Link first MIDI 2.0 Ready product: Roland A-88MKII https://www.midi.org/articles-old/roland-announces Issue URL : https://github.com/alsa-project/alsa-lib/issues/24 Repository URL: https://github.com/alsa-project/alsa-lib

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