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[alsa-devel] Applied "ASoC: txx9: use devm_snd_soc_register_component()" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: txx9: use devm_snd_soc_register_component() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fb77436a444e9836e6b1b0a457bc9c09cdce22f6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 7 Sep 2018 01:03:53 +0000 Subject: [PATCH] ASoC: txx9: use devm_snd_soc_register_component() Now we have devm_snd_soc_register_component(). Let's use it instead of snd_soc_register_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/txx9/txx9aclc-ac97.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e2ad00e3cae1..1cfca698ae4b 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -208,13 +208,12 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (err < 0) return err; - return snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, + return devm_snd_soc_register_component(&pdev->dev, &txx9aclc_ac97_component, &txx9aclc_ac97_dai, 1); } static int txx9aclc_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); return 0; } -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: q6core: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: q6core: Use kmemdup to replace kzalloc + memcpy has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 18fbe800e6066050ab6ae7751708da04975cdc22 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:19 +0800 Subject: [PATCH] ASoC: q6core: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to use the kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index 06f03a5fe9bd..ca1be7305524 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -105,12 +105,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) bytes = sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); - core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; - memcpy(core->fwk_version, data->payload, bytes); - core->fwk_version_supported = true; core->resp_received = true; @@ -124,12 +122,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - core->svc_version = kzalloc(len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; - memcpy(core->svc_version, data->payload, len); - core->get_version_supported = true; core->resp_received = true; -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ca92cc4636fdedf0d7ee88a5e50cd2b85c246a3b Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:20 +0800 Subject: [PATCH] ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..52a9915da0f5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, bc = (struct skl_algo_data *)sb->dobj.private; if (bc->set_params == SKL_PARAM_BIND) { - params = kzalloc(bc->max, GFP_KERNEL); + params = kmemdup(bc->params, bc->max, GFP_KERNEL); if (!params) return -ENOMEM; - memcpy(params, bc->params, bc->max); skl_fill_sink_instance_id(ctx, params, bc->max, mconfig); -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e36a1d0d249aa09f94d551cadf043a7f9f7fae00 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Mon, 10 Sep 2018 15:28:39 +0100 Subject: [PATCH] ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai snd_soc_dapm_new_dai may return an error pointer and currently this isn't checked for in dapm_connect_dai_link_widgets. Add code to check the return value and not add routes in that case. Fixes: 778ff5bb8689 ("ASoC: dapm: Move connection of CODEC to CODEC DAIs") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0dcdcc23dcfd..43983c69f6aa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4087,6 +4087,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, playback = snd_soc_dapm_new_dai(card, rtd, playback_cpu, codec); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + continue; + } snd_soc_dapm_add_path(&card->dapm, playback_cpu, playback, NULL, NULL); @@ -4099,7 +4106,9 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, snd_soc_dapm_add_path(&card->dapm, playback, codec, NULL, NULL); } + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI capture if widgets are valid */ codec = codec_dai->capture_widget; @@ -4108,6 +4117,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture = snd_soc_dapm_new_dai(card, rtd, codec, capture_cpu); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + continue; + } snd_soc_dapm_add_path(&card->dapm, capture, capture_cpu, NULL, NULL); -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: AMD: Fix simultaneous playback and capture on different channel" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: AMD: Fix simultaneous playback and capture on different channel has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8dcb0c90c691de5b79608d04ec7941ef9b3fee9c Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Mon, 10 Sep 2018 22:50:27 +0530 Subject: [PATCH] ASoC: AMD: Fix simultaneous playback and capture on different channel If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++----- sound/soc/amd/acp-pcm-dma.c | 8 ++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879cccbd2c0..717a017f0db6 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * 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); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static const struct snd_soc_ops cz_da7219_play_ops = { + .startup = cz_da7219_play_startup, + .shutdown = cz_da7219_shutdown, +}; + static const struct snd_soc_ops cz_da7219_cap_ops = { - .startup = cz_da7219_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index e359938e3d7e..8f3bc6e37f26 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -846,8 +846,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e8f4fa..dbbb1a85638d 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: pxa: switch to new ac97 bus support" to the asoc tree
by Mark Brown 11 Sep '18

11 Sep '18

The patch ASoC: pxa: switch to new ac97 bus support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1c8bc7b3de5e76cb89aacdc7be1475a028af505f Mon Sep 17 00:00:00 2001 From: Robert Jarzmik <robert.jarzmik(a)free.fr> Date: Sat, 25 Aug 2018 10:46:18 +0200 Subject: [PATCH] ASoC: pxa: switch to new ac97 bus support Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++-- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++------------------ 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 5fbd47a9177e..28867732a318 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -31,7 +31,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..29f577e6dfc0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -19,14 +19,13 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S select SND_PXA2XX_LIB diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..f8a3aa6c6d4e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.19.0.rc2

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[alsa-devel] [PATCH v2] ASoC: Fix UBSAN warning at snd_soc_get/put_volsw_sx()
by Rohit kumar 11 Sep '18

11 Sep '18

In functions snd_soc_get_volsw_sx() or snd_soc_put_volsw_sx(), if the result of (min + max) is negative, then fls() returns signed integer with value as 32. This leads to signed integer overflow as complete operation is considered as signed integer. UBSAN: Undefined behaviour in sound/soc/soc-ops.c:382:50 signed integer overflow: -2147483648 - 1 cannot be represented in type 'int' Call trace: [<ffffff852f746fe4>] __dump_stack lib/dump_stack.c:15 [inline] [<ffffff852f746fe4>] dump_stack+0xec/0x158 lib/dump_stack.c:51 [<ffffff852f7b5f3c>] ubsan_epilogue+0x18/0x50 lib/ubsan.c:164 [<ffffff852f7b6840>] handle_overflow+0xf8/0x130 lib/ubsan.c:195 [<ffffff852f7b68f0>] __ubsan_handle_sub_overflow+0x34/0x44 lib/ubsan.c:211 [<ffffff85307971a0>] snd_soc_get_volsw_sx+0x1a8/0x1f8 sound/soc/soc-ops.c:382 Typecast the operation to unsigned int to fix the issue. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Changes since v1: Addressed comments given by Takashi. sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 592efb3..f4dc3d4 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1U << (fls(min + max) - 1)) - 1; unsigned int val; int ret; @@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1U << (fls(min + max) - 1)) - 1; int err = 0; unsigned int val, val_mask, val2 = 0; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH] ASoC: Fix UBSAN warning at snd_soc_get/put_volsw_sx()
by Rohit kumar 11 Sep '18

11 Sep '18

In functions snd_soc_get_volsw_sx() or snd_soc_put_volsw_sx(), if the result of (min + max) is negative, then fls() returns signed integer with value as 32. This leads to signed integer overflow as complete operation is considered as signed integer. UBSAN: Undefined behaviour in sound/soc/soc-ops.c:382:50 signed integer overflow: -2147483648 - 1 cannot be represented in type 'int' Call trace: [<ffffff852f746fe4>] __dump_stack lib/dump_stack.c:15 [inline] [<ffffff852f746fe4>] dump_stack+0xec/0x158 lib/dump_stack.c:51 [<ffffff852f7b5f3c>] ubsan_epilogue+0x18/0x50 lib/ubsan.c:164 [<ffffff852f7b6840>] handle_overflow+0xf8/0x130 lib/ubsan.c:195 [<ffffff852f7b68f0>] __ubsan_handle_sub_overflow+0x34/0x44 lib/ubsan.c:211 [<ffffff85307971a0>] snd_soc_get_volsw_sx+0x1a8/0x1f8 sound/soc/soc-ops.c:382 Typecast the operation to unsigned int to fix the issue. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 592efb3..f8e3190 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -373,7 +373,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = ((unsigned int)(1 << (fls(min + max) - 1)) - 1); unsigned int val; int ret; @@ -418,7 +418,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - unsigned int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = ((unsigned int)(1 << (fls(min + max) - 1)) - 1); int err = 0; unsigned int val, val_mask, val2 = 0; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH - DMA engine v2 1/1] ALSA: SOC: DMA: increment buffer pointer atomically
by twischer＠de.adit-jv.com 11 Sep '18

11 Sep '18

From: Andreas Pape <apape(a)de.adit-jv.com> Setting pointer and afterwards check for wrap around leads to the possibility of returning the inconsistent pointer position. This patch increments buffer pointer atomically to avoid this issue. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Since commit: b7ae6f3 dmaengine implementation has been moved from ASoC to ALSA core. So change goes to sound/soc/soc-dmaengine-pcm.c, apply to sound/core/pcm_dmaengine.c now cherry picked from ADIT v3.14 e2f803f("ALSA: SOC: DMA: increment buffer pointer atomically") Signed-off-by: Veeraiyan Chidambaram <veeraiyan.chidambaram(a)in.bosch.com> (cherry picked from ADIT v4.1 commit 9e7eff10a38157db75f12e032fc378d3104a2812) Signed-off-by: Takashi Sakamoto <takashi.sakamoto(a)miraclelinux.com> (cherry picked from ADIT v4.9 commit 66c00aa1e7cd3b077fc2e5a45a4594b86246cf78) Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- v1 can be found in http://mailman.alsa-project.org/pipermail/alsa-devel/2016-November/115277.h… diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 8eb58c7..0a7ed85 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -139,12 +139,14 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_config_from_dai_data); static void dmaengine_pcm_dma_complete(void *arg) { + unsigned int new_pos; struct snd_pcm_substream *substream = arg; struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - prtd->pos += snd_pcm_lib_period_bytes(substream); - if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) - prtd->pos = 0; + new_pos = READ_ONCE(prtd->pos) + snd_pcm_lib_period_bytes(substream); + if (new_pos >= snd_pcm_lib_buffer_bytes(substream)) + new_pos = 0; + WRITE_ONCE(prtd->pos, new_pos); snd_pcm_period_elapsed(substream); } @@ -235,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); - return bytes_to_frames(substream->runtime, prtd->pos); + return bytes_to_frames(substream->runtime, READ_ONCE(prtd->pos)); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] ASoC: soc-core: use more generic method to find platform
by Kuninori Morimoto 11 Sep '18

11 Sep '18

Hi Mark Now, ALSA SoC is using snd_soc_dai_link_component for platform instead of legacy style (= platform_name/platform_of_node/platform). OTOH, CPU/Codec are finding its DAI by using common snd_soc_find_dai() function which uses snd_soc_dai_link_component. Then, of course, we want to use same style for platform. These patch do it. Kuninori Morimoto (3): ASoC: soc-core: manage platform name under snd_soc_init_platform() ASoC: soc-core: add snd_soc_find_component() ASoC: soc-core: find platform by using snd_soc_find_component() sound/soc/soc-core.c | 84 +++++++++++++++++++++++++++------------------------- 1 file changed, 43 insertions(+), 41 deletions(-) -- 2.7.4

1 4

[alsa-devel] [PATCH 0/2] sound: Use kmemdup to replace kzalloc + memcpy
by zhong jiang 11 Sep '18

11 Sep '18

I find the issue with the help of Coccinelle. zhong jiang (2): sound: q6core: Use kmemdup to replace kzalloc + memcpy sound: skl-topology: Use kmemdup to replace kzalloc + memcpy sound/soc/intel/skylake/skl-topology.c | 3 +-- sound/soc/qcom/qdsp6/q6core.c | 8 ++------ 2 files changed, 3 insertions(+), 8 deletions(-) -- 1.7.12.4

2 4

[alsa-devel] [PATCH 00/16] ASoC: use devm_snd_soc_register_component()
by Kuninori Morimoto 11 Sep '18

11 Sep '18

Hi Mark We have devm_ version of component register function. But, few drivers still doesn't use it. Let's convert it. Kuninori Morimoto (16): ASoC: atmel_ssc_dai: use devm_snd_soc_register_component() ASoC: au1x: ac97c: use devm_snd_soc_register_component() ASoC: bcm: use devm_snd_soc_register_component() ASoC: cirrus: use devm_snd_soc_register_component() ASoC: hdac_hda: use devm_snd_soc_register_component() ASoC: rt5668: use devm_snd_soc_register_component() ASoC: tscs454: use devm_snd_soc_register_component() ASoC: davinci: use devm_snd_soc_register_component() ASoC: fsl: use devm_snd_soc_register_component() ASoC: nuc900: use devm_snd_soc_register_component() ASoC: omap: use devm_snd_soc_register_component() ASoC: pxa: use devm_snd_soc_register_component() ASoC: sh: use devm_snd_soc_register_component() ASoC: tegra: use devm_snd_soc_register_component() ASoC: txx9: use devm_snd_soc_register_component() ASoC: ux500: use devm_snd_soc_register_component() sound/soc/atmel/atmel_ssc_dai.c | 13 +++---------- sound/soc/au1x/ac97c.c | 4 +--- sound/soc/au1x/i2sc.c | 4 +--- sound/soc/au1x/psc-ac97.c | 4 +--- sound/soc/au1x/psc-i2s.c | 4 +--- sound/soc/bcm/cygnus-ssp.c | 13 ++++--------- sound/soc/cirrus/ep93xx-ac97.c | 8 ++------ sound/soc/cirrus/ep93xx-i2s.c | 7 ++----- sound/soc/codecs/hdac_hda.c | 3 +-- sound/soc/codecs/rt5668.c | 4 +--- sound/soc/codecs/tscs454.c | 2 +- sound/soc/davinci/davinci-i2s.c | 8 ++------ sound/soc/fsl/imx-ssi.c | 8 ++------ sound/soc/fsl/mpc5200_psc_ac97.c | 3 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +-- sound/soc/nuc900/nuc900-ac97.c | 4 +--- sound/soc/omap/omap-hdmi-audio.c | 4 +--- sound/soc/pxa/pxa2xx-ac97.c | 3 +-- sound/soc/sh/hac.c | 3 +-- sound/soc/tegra/tegra20_ac97.c | 7 ++----- sound/soc/tegra/tegra20_i2s.c | 7 ++----- sound/soc/tegra/tegra20_spdif.c | 7 ++----- sound/soc/tegra/tegra30_i2s.c | 7 ++----- sound/soc/txx9/txx9aclc-ac97.c | 3 +-- sound/soc/ux500/ux500_msp_dai.c | 10 ++-------- 25 files changed, 39 insertions(+), 104 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: dapm: Add missing return value check for snd_soc_dapm_new_dai
by Charles Keepax 10 Sep '18

10 Sep '18

snd_soc_dapm_new_dai may return an error pointer and currently this isn't checked for in dapm_connect_dai_link_widgets. Add code to check the return value and not add routes in that case. Fixes: 778ff5bb8689 ("ASoC: dapm: Move connection of CODEC to CODEC DAIs") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0dcdcc23dcfde..43983c69f6aad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4087,6 +4087,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, playback = snd_soc_dapm_new_dai(card, rtd, playback_cpu, codec); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + continue; + } snd_soc_dapm_add_path(&card->dapm, playback_cpu, playback, NULL, NULL); @@ -4099,7 +4106,9 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, snd_soc_dapm_add_path(&card->dapm, playback, codec, NULL, NULL); } + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* connect BE DAI capture if widgets are valid */ codec = codec_dai->capture_widget; @@ -4108,6 +4117,13 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture = snd_soc_dapm_new_dai(card, rtd, codec, capture_cpu); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + continue; + } snd_soc_dapm_add_path(&card->dapm, capture, capture_cpu, NULL, NULL); -- 2.11.0

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[alsa-devel] Applied "ASoC: AMD: Fix simultaneous playback and capture on different channel" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: AMD: Fix simultaneous playback and capture on different channel has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8dcb0c90c691de5b79608d04ec7941ef9b3fee9c Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Mon, 10 Sep 2018 22:50:27 +0530 Subject: [PATCH] ASoC: AMD: Fix simultaneous playback and capture on different channel If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++----- sound/soc/amd/acp-pcm-dma.c | 8 ++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879cccbd2c0..717a017f0db6 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * 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); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static const struct snd_soc_ops cz_da7219_play_ops = { + .startup = cz_da7219_play_startup, + .shutdown = cz_da7219_shutdown, +}; + static const struct snd_soc_ops cz_da7219_cap_ops = { - .startup = cz_da7219_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index e359938e3d7e..8f3bc6e37f26 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -846,8 +846,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e8f4fa..dbbb1a85638d 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- 2.19.0.rc1

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[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 10 Sep '18

10 Sep '18

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 5fbd47a9177e..28867732a318 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -31,7 +31,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..29f577e6dfc0 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -19,14 +19,13 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S select SND_PXA2XX_LIB diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9f779657bc86..f8a3aa6c6d4e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -233,6 +226,9 @@ MODULE_DEVICE_TABLE(of, pxa2xx_ac97_dt_ids); static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -245,10 +241,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -259,8 +259,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: AMD: Fix simultaneous playback and capture on different channel
by Akshu Agrawal 10 Sep '18

10 Sep '18

If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having lock between startup and prepare. This ensures no other codec startup gets called between a codec's startup(where channel info is set) and hw_params(where channel info is read). Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879ccc..b98ffbc 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,6 +47,7 @@ static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; +static struct mutex instance_lock; extern int bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -150,6 +151,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); @@ -160,6 +162,12 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_da7219_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static int cz_max_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -177,6 +185,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -186,6 +195,12 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_max_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static int cz_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -203,6 +218,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,6 +240,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); @@ -234,24 +251,34 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_dmic_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_startup, .shutdown = cz_da7219_shutdown, + .prepare = cz_da7219_prepare, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .prepare = cz_max_prepare, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .prepare = cz_dmic_prepare, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .prepare = cz_dmic_prepare, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { @@ -409,6 +436,7 @@ static int cz_probe(struct platform_device *pdev) card = &cz_card; cz_card.dev = &pdev->dev; platform_set_drvdata(pdev, card); + mutex_init(&instance_lock); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); if (ret) { -- 1.9.1

3 4

[alsa-devel] [PATCH v2] ASoC: AMD: Fix simultaneous playback and capture on different channel
by Akshu Agrawal 10 Sep '18

10 Sep '18

If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++++++++------- sound/soc/amd/acp-pcm-dma.c | 8 ++++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 55d7f61..0d97d00 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static void da7219_clk_disable(void) .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * 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); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static const struct snd_soc_ops cz_da7219_play_ops = { + .startup = cz_da7219_play_startup, + .shutdown = cz_da7219_shutdown, +}; + static const struct snd_soc_ops cz_da7219_cap_ops = { - .startup = cz_da7219_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b43f0a1..b0e245c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -862,8 +862,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e..dbbb1a8 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: emu10k1: fix possible info leak to userspace on SNDRV_EMU10K1_IOCTL_INFO
by Takashi Iwai 10 Sep '18

10 Sep '18

From: Willy Tarreau <w(a)1wt.eu> snd_emu10k1_fx8010_ioctl(SNDRV_EMU10K1_IOCTL_INFO) allocates memory using kmalloc() and partially fills it by calling snd_emu10k1_fx8010_info() before returning the resulting structure to userspace, leaving uninitialized holes. Let's just use kzalloc() here. BugLink: http://blog.infosectcbr.com.au/2018/09/linux-kernel-infoleaks.html Signed-off-by: Willy Tarreau <w(a)1wt.eu> Cc: Jann Horn <jannh(a)google.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/emufx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 90713741c2dc..6ebe817801ea 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2540,7 +2540,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un emu->support_tlv = 1; return put_user(SNDRV_EMU10K1_VERSION, (int __user *)argp); case SNDRV_EMU10K1_IOCTL_INFO: - info = kmalloc(sizeof(*info), GFP_KERNEL); + info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; snd_emu10k1_fx8010_info(emu, info); -- 2.18.0

1 1

[alsa-devel] RaspberryPi vc4-hdmi softvol for alsamixer
by Andreas Müller 10 Sep '18

10 Sep '18

Hi, This is my first post here and guess what I am looking for support :) Up to now I used audioinjector and hifiberry on RaspberryPi and that worked without issues (from alsa point of view). Now I would like to select vc4hdmi on alsamixer but that complains for missing controls. I think vc4hdmi does not support hard volume setting - like some other devices. So I read here and there and found softvol. To test I modified vc4-hdmi.conf shipped with alsal-iib from: # default with dmix vc4-hdmi.pcm.default { @args [ CARD ] @args.CARD { type string } type asym playback.pcm { type plug slave.pcm { @func concat strings [ "dmix:" $CARD ] } } } to (stolen from PC-Speaker.conf) # default with dmix vc4-hdmi.pcm.default { @args [ CARD ] @args.CARD { type string } type asym playback.pcm { type plug slave.pcm { type softvol control { name "Master Playback Volume" card $CARD } slave.pcm { @func concat strings [ "dmix:" $CARD ] } } } } after reboot nothing changes! I build my images with Openembedded/Yocto and from my experience it is possible that there is some misconfiguration / path issue causes problems. I know the card works with the help of pulseaudio. So now my questions: * Would you expect that my modification of vc4-hdmi.conf adds softvol control to vc4hdmi? * If yes: The only information I have currently: It does not work as expected. So where can I ask alsa for some feedback what went wrong (log-files/dmesg/journal...)? Thanks in advance Andreas

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix a couple uninitialized ret variables
by Dan Carpenter 10 Sep '18

10 Sep '18

Smatch complains that these variables could be uninitialized. The first one in snd_soc_dai_link_event() is probably a false positive, because probably we know the lists are not empty. I would normally ignore the warning, but GCC complains here as well so I just silenced the warning. The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it leads to a bogus dereference in the caller. Fixes: 3bbf5d34fd4a ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f4edcd19e02..0dcdcc23dcfd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3623,7 +3623,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; config = rtd->dai_link->params + rtd->params_select; @@ -3938,8 +3938,10 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) + if (IS_ERR(w)) { + ret = PTR_ERR(w); goto outfree_kcontrol_news; + } w->priv = rtd;

2 1

[alsa-devel] Applied "ASoC: q6core: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: q6core: Use kmemdup to replace kzalloc + memcpy has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 18fbe800e6066050ab6ae7751708da04975cdc22 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:19 +0800 Subject: [PATCH] ASoC: q6core: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to use the kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index 06f03a5fe9bd..ca1be7305524 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -105,12 +105,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) bytes = sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); - core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; - memcpy(core->fwk_version, data->payload, bytes); - core->fwk_version_supported = true; core->resp_received = true; @@ -124,12 +122,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - core->svc_version = kzalloc(len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; - memcpy(core->svc_version, data->payload, len); - core->get_version_supported = true; core->resp_received = true; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ca92cc4636fdedf0d7ee88a5e50cd2b85c246a3b Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:20 +0800 Subject: [PATCH] ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..52a9915da0f5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, bc = (struct skl_algo_data *)sb->dobj.private; if (bc->set_params == SKL_PARAM_BIND) { - params = kzalloc(bc->max, GFP_KERNEL); + params = kmemdup(bc->params, bc->max, GFP_KERNEL); if (!params) return -ENOMEM; - memcpy(params, bc->params, bc->max); skl_fill_sink_instance_id(ctx, params, bc->max, mconfig); -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH] ASoC: qdsp6: q6asm-dai: clean up a return
by Dan Carpenter 10 Sep '18

10 Sep '18

Smatch complains that if both "psubstream" and "csubstream" are NULL then "ret" is uninitialized. That probably can't happen, but it's cleaner to just return zero anyway so let's do that. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9db9a2944ef2..c75fab38905d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) } } - return ret; + return 0; } static void q6asm_dai_pcm_free(struct snd_pcm *pcm)

3 2

[alsa-devel] [PATCH] ALSA: fireface: fix memory leak in ff400_switch_fetching_mode()
by Takashi Sakamoto 10 Sep '18

10 Sep '18

An allocated memory forgets to be released. Fixes: 76fdb3a9e13 ('ALSA: fireface: add support for Fireface 400') Cc: <stable(a)vger.kernel.org> # 4.12+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-ff400.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index ad7a0a32557d..64c3cb0fb926 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -146,6 +146,7 @@ static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) { __le32 *reg; int i; + int err; reg = kcalloc(18, sizeof(__le32), GFP_KERNEL); if (reg == NULL) @@ -163,9 +164,11 @@ static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) reg[i] = cpu_to_le32(0x00000001); } - return snd_fw_transaction(ff->unit, TCODE_WRITE_BLOCK_REQUEST, - FF400_FETCH_PCM_FRAMES, reg, - sizeof(__le32) * 18, 0); + err = snd_fw_transaction(ff->unit, TCODE_WRITE_BLOCK_REQUEST, + FF400_FETCH_PCM_FRAMES, reg, + sizeof(__le32) * 18, 0); + kfree(reg); + return err; } static void ff400_dump_sync_status(struct snd_ff *ff, -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: bebob: use address returned by kmalloc() instead of kernel stack for streaming DMA mapping
by Takashi Sakamoto 10 Sep '18

10 Sep '18

When executing 'fw_run_transaction()' with 'TCODE_WRITE_BLOCK_REQUEST', an address of 'payload' argument is used for streaming DMA mapping by 'firewire_ohci' module if 'size' argument is larger than 8 byte. Although in this case the address should not be on kernel stack, current implementation of ALSA bebob driver uses data in kernel stack for a cue to boot M-Audio devices. This often brings unexpected result, especially for a case of CONFIG_VMAP_STACK=y. This commit fixes the bug. Reference: https://bugzilla.kernel.org/show_bug.cgi?id=201021 Reference: https://forum.manjaro.org/t/firewire-m-audio-410-driver-wont-load-firmware/… Fixes: a2b2a7798fb6('ALSA: bebob: Send a cue to load firmware for M-Audio Firewire series') Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_maudio.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index bd55620c6a47..0c5a4cbb99ba 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -96,17 +96,13 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) struct fw_device *device = fw_parent_device(unit); int err, rcode; u64 date; - __le32 cues[3] = { - cpu_to_le32(MAUDIO_BOOTLOADER_CUE1), - cpu_to_le32(MAUDIO_BOOTLOADER_CUE2), - cpu_to_le32(MAUDIO_BOOTLOADER_CUE3) - }; + __le32 *cues; /* check date of software used to build */ err = snd_bebob_read_block(unit, INFO_OFFSET_SW_DATE, &date, sizeof(u64)); if (err < 0) - goto end; + return err; /* * firmware version 5058 or later has date later than "20070401", but * 'date' is not null-terminated. @@ -114,20 +110,28 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) if (date < 0x3230303730343031LL) { dev_err(&unit->device, "Use firmware version 5058 or later\n"); - err = -ENOSYS; - goto end; + return -ENXIO; } + cues = kmalloc_array(3, sizeof(*cues), GFP_KERNEL); + if (!cues) + return -ENOMEM; + + cues[0] = cpu_to_le32(MAUDIO_BOOTLOADER_CUE1); + cues[1] = cpu_to_le32(MAUDIO_BOOTLOADER_CUE2); + cues[2] = cpu_to_le32(MAUDIO_BOOTLOADER_CUE3); + rcode = fw_run_transaction(device->card, TCODE_WRITE_BLOCK_REQUEST, device->node_id, device->generation, device->max_speed, BEBOB_ADDR_REG_REQ, - cues, sizeof(cues)); + cues, 3 * sizeof(*cues)); + kfree(cues); if (rcode != RCODE_COMPLETE) { dev_err(&unit->device, "Failed to send a cue to load firmware\n"); err = -EIO; } -end: + return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: pcm: Update hardware pointer before start capture
by Ricardo Biehl Pasquali 10 Sep '18

10 Sep '18

This ensures the transfer loop won't waste a run to read the few frames (if any) between start and hw_ptr update. It will wait for the next interrupt with wait_for_avail(). Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- sound/core/pcm_lib.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7f71c2449..40013b26f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2172,6 +2172,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (err < 0) goto _end_unlock; + runtime->twake = runtime->control->avail_min ? : 1; + if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_update_hw_ptr(substream); + if (!is_playback && runtime->status->state == SNDRV_PCM_STATE_PREPARED) { if (size >= runtime->start_threshold) { @@ -2185,10 +2189,8 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, } } - runtime->twake = runtime->control->avail_min ? : 1; - if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) - snd_pcm_update_hw_ptr(substream); avail = snd_pcm_avail(substream); + while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t cont; -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: core: add support to card rebind
by Srinivas Kandagatla 10 Sep '18

10 Sep '18

Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi Mark, This time I tried doing this without component framework, by maintaining list of ubind card in ASoC core that are still not unregistered. This list is tried to bind for every successful add of component. It looks much better than the component framework one! thanks, srini include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 86 ++++++++++++++++++++++++++++++-------------- 2 files changed, 61 insertions(+), 27 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a4915148f739..d9a8aea213cf 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1092,6 +1092,7 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head component_dev_list; + struct list_head list; struct list_head widgets; struct list_head paths; @@ -1359,6 +1360,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); + INIT_LIST_HEAD(&card->list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6d33634b934b..795796978fc8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,7 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -2678,6 +2679,34 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int j; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2687,7 +2716,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2719,28 +2747,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2750,12 +2773,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3101,7 +3120,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3141,6 +3160,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3168,6 +3199,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; -- 2.18.0

2 2

Re: [alsa-devel] [PATCH] ASoC: max98373: usleep_range() needs include/delay.h
by Mark Brown 07 Sep '18

07 Sep '18

On Thu, Sep 06, 2018 at 05:27:28PM -0700, Grant Grundler wrote: > Commit ca917f9fe1a0fab added use of usleep_range() but not > the corresponding "include <linux/delay.h>". The result is > with Chrome OS won't build because warnings are forced > to be errors: > mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration] > usleep_range(10000, 11000); > ^ Note that this isn't causing a warning upstream, presumably due to an implicit inclusion that isn't present in the v4.4 kernel that you appear to be using, or gets missed due to config differences.

2 4

[alsa-devel] [bug report] ASoC: dapm: Move connection of CODEC to CODEC DAIs
by Dan Carpenter 07 Sep '18

07 Sep '18

Hello Charles Keepax, The patch 778ff5bb8689: "ASoC: dapm: Move connection of CODEC to CODEC DAIs" from Sep 5, 2018, leads to the following static checker warning: sound/soc/soc-dapm.c:4089 dapm_connect_dai_link_widgets() error: 'playback' dereferencing possible ERR_PTR() sound/soc/soc-dapm.c:4110 dapm_connect_dai_link_widgets() error: 'capture' dereferencing possible ERR_PTR() sound/soc/soc-dapm.c 4077 4078 for_each_rtd_codec_dai(rtd, i, codec_dai) { 4079 4080 /* connect BE DAI playback if widgets are valid */ 4081 codec = codec_dai->playback_widget; 4082 4083 if (playback_cpu && codec) { 4084 if (!playback) { 4085 playback = snd_soc_dapm_new_dai(card, rtd, 4086 playback_cpu, 4087 codec); 4088 4089 snd_soc_dapm_add_path(&card->dapm, playback_cpu, 4090 playback, NULL, NULL); ^^^^^^^^ snd_soc_dapm_new_dai() returns error pointers. 4091 } 4092 4093 dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", 4094 cpu_dai->component->name, playback_cpu->name, 4095 codec_dai->component->name, codec->name); 4096 4097 snd_soc_dapm_add_path(&card->dapm, playback, codec, 4098 NULL, NULL); 4099 } 4100 4101 /* connect BE DAI capture if widgets are valid */ 4102 codec = codec_dai->capture_widget; 4103 4104 if (codec && capture_cpu) { 4105 if (!capture) { 4106 capture = snd_soc_dapm_new_dai(card, rtd, 4107 codec, 4108 capture_cpu); 4109 4110 snd_soc_dapm_add_path(&card->dapm, capture, 4111 capture_cpu, NULL, NULL); ^^^^^^^^^^^ 4112 } 4113 4114 dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", 4115 codec_dai->component->name, codec->name, 4116 cpu_dai->component->name, capture_cpu->name); 4117 4118 snd_soc_dapm_add_path(&card->dapm, codec, capture, 4119 NULL, NULL); regards, dan carpenter

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: hdac_hdmi: Limit sampling rates at dai creation
by Yong Zhi 07 Sep '18

07 Sep '18

Playback of 44.1Khz contents with HDMI plugged returns "Invalid pipe config" because HDMI paths in the FW topology are configured to operate at 48Khz. This patch filters out sampling rates not supported at hdac_hdmi_create_dais() to let user space SRC to do the converting. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> --- v2: - Add Reviewed-by tag. - Use (and tested) Takashi's sample code in his v1 review. sound/soc/codecs/hdac_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..590dc1e33a60 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1410,6 +1410,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdev->dev, dai_name, GFP_KERNEL); -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: max98373: usleep_range() needs include/delay.h" to the asoc tree
by Mark Brown 07 Sep '18

07 Sep '18

The patch ASoC: max98373: usleep_range() needs include/delay.h has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3004136b90bedc9e254ff659adb7a60299e9495e Mon Sep 17 00:00:00 2001 From: Grant Grundler <grundler(a)chromium.org> Date: Thu, 6 Sep 2018 17:27:28 -0700 Subject: [PATCH] ASoC: max98373: usleep_range() needs include/delay.h Commit ca917f9fe1a0fab added use of usleep_range() but not the corresponding "include <linux/delay.h>". The result is with Chrome OS won't build because warnings are forced to be errors: mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration] usleep_range(10000, 11000); ^ Including delay.h "fixes" this. Signed-off-by: Grant Grundler <grundler(a)chromium.org> Reviewed-by: Benson Leung <bleung(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 1093f766d0d2..d6868c9a9ce6 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -2,6 +2,7 @@ // Copyright (c) 2017, Maxim Integrated #include <linux/acpi.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/regmap.h> -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH] ASoC: cs4265: Add a MIC pre. switch
by Matt Flax 07 Sep '18

07 Sep '18

This patch adds a MIC preamp enable switch as a SOC_SINGLE. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d9eebf6a..03476f93 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -161,6 +161,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), + SOC_SINGLE("MIC Preamp", CS4265_PWRCTL, 3, 1, 1), }; static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { -- 2.17.1

2 4

[alsa-devel] [PATCH v2] ASoC: rsnd: Add device tree binding for r8a77990
by Yoshihiro Kaneko 07 Sep '18

07 Sep '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> This patch adds the device tree binding of the r8a77990 SoC. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> --- This patch is based on the devel branch of Simon Horman's renesas tree. v2 [Yoshihiro Kaneko] - Rebased Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 9e764270..58971a6 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -353,6 +353,7 @@ Required properties: - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - "renesas,rcar_sound-r8a77965" (R-Car M3-N) + - "renesas,rcar_sound-r8a77990" (R-Car E3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: dapm: Avoid uninitialised variable warning
by Charles Keepax 07 Sep '18

07 Sep '18

Commit 4a75aae17b2a ("ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links") adds loops that iterate over multiple CODECs in snd_soc_dai_link_event. This also introduced a compiler warning for a potentially uninitialised variable in the case no CODECs are present. This should never be the case as the DAI link must by definition contain at least 1 CODEC however probably best to avoid the compiler warning by initialising ret to zero. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f4edcd19e022..e496bc642c990 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3623,7 +3623,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; config = rtd->dai_link->params + rtd->params_select; -- 2.11.0

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Re: [alsa-devel] [PATCH] ASoC: max98373: usleep_range() needs include/delay.h
by Benson Leung 07 Sep '18

07 Sep '18

Hi Grant, On Thu, Sep 06, 2018 at 05:27:28PM -0700, Grant Grundler wrote: > Commit ca917f9fe1a0fab added use of usleep_range() but not > the corresponding "include <linux/delay.h>". The result is > with Chrome OS won't build because warnings are forced > to be errors: > mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration] > usleep_range(10000, 11000); > ^ > > Including delay.h "fixes" this. > > Signed-off-by: Grant Grundler <grundler(a)chromium.org> Reviewed-by: Benson Leung <bleung(a)chromium.org> Thanks! -- Benson Leung Staff Software Engineer Chrome OS Kernel Google Inc. bleung(a)google.com Chromium OS Project bleung(a)chromium.org

1 0

[alsa-devel] [asoc:for-4.20 399/404] sound/soc/soc-dapm.c:3768:9: warning: 'ret' may be used uninitialized in this function
by kbuild test robot 06 Sep '18

06 Sep '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 9ab708aef61f5620113269a9d1bdb1543d1207d0 commit: 4a75aae17b2a802a7267206414050408392c374c [399/404] ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links config: x86_64-randconfig-x006-201835 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 4a75aae17b2a802a7267206414050408392c374c # save the attached .config to linux build tree make ARCH=x86_64 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_power_one_widget Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_check_dynamic_path Cyclomatic Complexity 2 sound/soc/soc-dapm.c:soc_dapm_update_bits Cyclomatic Complexity 15 sound/soc/soc-dapm.c:dapm_widget_update Cyclomatic Complexity 3 sound/soc/soc-dapm.c:pop_dbg Cyclomatic Complexity 12 sound/soc/soc-dapm.c:dapm_seq_check_event Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_free_path Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_kcontrol_free Cyclomatic Complexity 2 sound/soc/soc-dapm.c:pop_wait Cyclomatic Complexity 13 sound/soc/soc-dapm.c:dapm_seq_run_coalesced Cyclomatic Complexity 42 sound/soc/soc-dapm.c:dapm_seq_run Cyclomatic Complexity 13 sound/soc/soc-dapm.c:dapm_widget_show_component Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_widget_show Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_wcache_lookup Cyclomatic Complexity 2 sound/soc/soc-dapm.c:soc_dapm_read Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_connect_mux Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_set_mixer_path_status Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_connect_mixer Cyclomatic Complexity 17 sound/soc/soc-dapm.c:dapm_update_widget_flags Cyclomatic Complexity 22 sound/soc/soc-dapm.c:snd_soc_dapm_add_path Cyclomatic Complexity 19 sound/soc/soc-dapm.c:snd_soc_dapm_add_route Cyclomatic Complexity 10 sound/soc/soc-dapm.c:snd_soc_dapm_del_route Cyclomatic Complexity 7 sound/soc/soc-dapm.c:dapm_find_widget Cyclomatic Complexity 11 sound/soc/soc-dapm.c:snd_soc_dapm_weak_route Cyclomatic Complexity 4 sound/soc/soc-dapm.c:snd_soc_dapm_set_pin Cyclomatic Complexity 4 sound/soc/soc-dapm.c:dapm_new_dai_link Cyclomatic Complexity 9 sound/soc/soc-dapm.c:dapm_is_shared_kcontrol Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_kcontrol_add_widget Cyclomatic Complexity 17 sound/soc/soc-dapm.c:dapm_widget_power_read_file Cyclomatic Complexity 2 sound/soc/soc-dapm.c:soc_dapm_test_bits Cyclomatic Complexity 8 sound/soc/soc-dapm.c:dapm_supply_check_power Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_free_kcontrol Cyclomatic Complexity 42 sound/soc/soc-dapm.c:snd_soc_dai_link_event Cyclomatic Complexity 2 include/linux/device.h:devm_kmalloc_array Cyclomatic Complexity 1 include/linux/device.h:devm_kcalloc Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_alloc_kcontrol Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_debugfs_cleanup Cyclomatic Complexity 7 sound/soc/soc-dapm.c:dapm_mark_endpoints_dirty Cyclomatic Complexity 1 sound/soc/soc-dapm.c:dapm_kcontrol_get_value Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_kcontrol_widget Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_kcontrol_dapm Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_force_bias_level Cyclomatic Complexity 9 sound/soc/soc-dapm.c:snd_soc_dapm_set_bias_level Cyclomatic Complexity 10 sound/soc/soc-dapm.c:dapm_pre_sequence_async Cyclomatic Complexity 11 sound/soc/soc-dapm.c:dapm_post_sequence_async Cyclomatic Complexity 47 sound/soc/soc-dapm.c:dapm_power_widgets Cyclomatic Complexity 9 sound/soc/soc-dapm.c:soc_dapm_mux_update_power Cyclomatic Complexity 9 sound/soc/soc-dapm.c:soc_dapm_mixer_update_power Cyclomatic Complexity 3 sound/soc/soc-dapm.c:soc_dapm_stream_event Cyclomatic Complexity 9 sound/soc/soc-dapm.c:soc_dapm_shutdown_dapm Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_dai_get_connected_widgets Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_regulator_event Cyclomatic Complexity 4 sound/soc/soc-dapm.c:dapm_pinctrl_event Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_clock_event Cyclomatic Complexity 4 sound/soc/soc-dapm.c:snd_soc_dapm_debugfs_init Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_mux_update_power Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_mixer_update_power Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_free_widget Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_reset_cache Cyclomatic Complexity 6 sound/soc/soc-dapm.c:dapm_free_widgets Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_sync_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_sync Cyclomatic Complexity 4 sound/soc/soc-dapm.c:snd_soc_dapm_add_routes Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_del_routes Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_weak_routes Cyclomatic Complexity 11 sound/soc/soc-dapm.c:snd_soc_dapm_get_volsw Cyclomatic Complexity 11 sound/soc/soc-dapm.c:snd_soc_dapm_put_volsw Cyclomatic Complexity 5 sound/soc/soc-dapm.c:snd_soc_dapm_get_enum_double Cyclomatic Complexity 8 sound/soc/soc-dapm.c:snd_soc_dapm_put_enum_double Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_info_pin_switch Cyclomatic Complexity 24 sound/soc/soc-dapm.c:snd_soc_dapm_new_control_unlocked Cyclomatic Complexity 13 sound/soc/soc-dapm.c:dapm_kcontrol_data_alloc Cyclomatic Complexity 14 sound/soc/soc-dapm.c:dapm_create_or_share_kcontrol Cyclomatic Complexity 9 sound/soc/soc-dapm.c:dapm_new_mixer Cyclomatic Complexity 10 sound/soc/soc-dapm.c:dapm_new_mux Cyclomatic Complexity 3 sound/soc/soc-dapm.c:dapm_new_pga Cyclomatic Complexity 13 sound/soc/soc-dapm.c:snd_soc_dapm_new_widgets Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_new_dai Cyclomatic Complexity 8 sound/soc/soc-dapm.c:dapm_connect_dai_link_widgets Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_new_control Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_new_controls Cyclomatic Complexity 6 sound/soc/soc-dapm.c:snd_soc_dapm_new_dai_widgets Cyclomatic Complexity 14 sound/soc/soc-dapm.c:snd_soc_dapm_link_dai_widgets Cyclomatic Complexity 5 sound/soc/soc-dapm.c:snd_soc_dapm_connect_dai_link_widgets Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_stream_event Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_enable_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_enable_pin Cyclomatic Complexity 3 sound/soc/soc-dapm.c:snd_soc_dapm_force_enable_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_force_enable_pin Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_disable_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_disable_pin Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_put_pin_switch Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_nc_pin_unlocked Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_nc_pin Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_get_pin_status Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_get_pin_switch Cyclomatic Complexity 2 sound/soc/soc-dapm.c:snd_soc_dapm_ignore_suspend Cyclomatic Complexity 1 sound/soc/soc-dapm.c:snd_soc_dapm_free Cyclomatic Complexity 7 sound/soc/soc-dapm.c:snd_soc_dapm_shutdown sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event': >> sound/soc/soc-dapm.c:3768:9: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] return ret; ^~~ vim +/ret +3768 sound/soc/soc-dapm.c 4ba1327a Mark Brown 2008-05-13 3613 c74184ed Mark Brown 2012-04-04 3614 static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, c74184ed Mark Brown 2012-04-04 3615 struct snd_kcontrol *kcontrol, int event) c74184ed Mark Brown 2012-04-04 3616 { 4a75aae1 Charles Keepax 2018-09-05 3617 struct snd_soc_dapm_path *path; c74184ed Mark Brown 2012-04-04 3618 struct snd_soc_dai *source, *sink; 249dc495 Charles Keepax 2018-08-15 3619 struct snd_soc_pcm_runtime *rtd = w->priv; c6615082 Nikesh Oswal 2015-02-02 3620 const struct snd_soc_pcm_stream *config = w->params + w->params_select; c74184ed Mark Brown 2012-04-04 3621 struct snd_pcm_substream substream; 9747cec2 Mark Brown 2012-04-26 3622 struct snd_pcm_hw_params *params = NULL; 8053f216 Nicolin Chen 2016-07-26 3623 struct snd_pcm_runtime *runtime = NULL; 3ba66feb Takashi Iwai 2018-07-25 3624 unsigned int fmt; c74184ed Mark Brown 2012-04-04 3625 int ret; c74184ed Mark Brown 2012-04-04 3626 bf4edea8 Takashi Iwai 2013-11-07 3627 if (WARN_ON(!config) || a3423b02 Lars-Peter Clausen 2015-08-11 3628 WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || a3423b02 Lars-Peter Clausen 2015-08-11 3629 list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) bf4edea8 Takashi Iwai 2013-11-07 3630 return -EINVAL; c74184ed Mark Brown 2012-04-04 3631 c74184ed Mark Brown 2012-04-04 3632 /* Be a little careful as we don't want to overflow the mask array */ c74184ed Mark Brown 2012-04-04 3633 if (config->formats) { c74184ed Mark Brown 2012-04-04 3634 fmt = ffs(config->formats) - 1; c74184ed Mark Brown 2012-04-04 3635 } else { 30a6a1a4 Liam Girdwood 2012-11-19 3636 dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", c74184ed Mark Brown 2012-04-04 3637 config->formats); c74184ed Mark Brown 2012-04-04 3638 fmt = 0; c74184ed Mark Brown 2012-04-04 3639 } c74184ed Mark Brown 2012-04-04 3640 c74184ed Mark Brown 2012-04-04 3641 /* Currently very limited parameter selection */ 9747cec2 Mark Brown 2012-04-26 3642 params = kzalloc(sizeof(*params), GFP_KERNEL); 9747cec2 Mark Brown 2012-04-26 3643 if (!params) { 9747cec2 Mark Brown 2012-04-26 3644 ret = -ENOMEM; 9747cec2 Mark Brown 2012-04-26 3645 goto out; 9747cec2 Mark Brown 2012-04-26 3646 } 9747cec2 Mark Brown 2012-04-26 3647 snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); c74184ed Mark Brown 2012-04-04 3648 9747cec2 Mark Brown 2012-04-26 3649 hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = c74184ed Mark Brown 2012-04-04 3650 config->rate_min; 9747cec2 Mark Brown 2012-04-26 3651 hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = c74184ed Mark Brown 2012-04-04 3652 config->rate_max; c74184ed Mark Brown 2012-04-04 3653 9747cec2 Mark Brown 2012-04-26 3654 hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min c74184ed Mark Brown 2012-04-04 3655 = config->channels_min; 9747cec2 Mark Brown 2012-04-26 3656 hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max c74184ed Mark Brown 2012-04-04 3657 = config->channels_max; c74184ed Mark Brown 2012-04-04 3658 c74184ed Mark Brown 2012-04-04 3659 memset(&substream, 0, sizeof(substream)); c74184ed Mark Brown 2012-04-04 3660 8053f216 Nicolin Chen 2016-07-26 3661 /* Allocate a dummy snd_pcm_runtime for startup() and other ops() */ 8053f216 Nicolin Chen 2016-07-26 3662 runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); 8053f216 Nicolin Chen 2016-07-26 3663 if (!runtime) { 8053f216 Nicolin Chen 2016-07-26 3664 ret = -ENOMEM; 8053f216 Nicolin Chen 2016-07-26 3665 goto out; 8053f216 Nicolin Chen 2016-07-26 3666 } 8053f216 Nicolin Chen 2016-07-26 3667 substream.runtime = runtime; 249dc495 Charles Keepax 2018-08-15 3668 substream.private_data = rtd; 8053f216 Nicolin Chen 2016-07-26 3669 c74184ed Mark Brown 2012-04-04 3670 switch (event) { c74184ed Mark Brown 2012-04-04 3671 case SND_SOC_DAPM_PRE_PMU: c74184ed Mark Brown 2012-04-04 3672 substream.stream = SNDRV_PCM_STREAM_CAPTURE; 4a75aae1 Charles Keepax 2018-09-05 3673 snd_soc_dapm_widget_for_each_source_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3674 source = path->source->priv; 4a75aae1 Charles Keepax 2018-09-05 3675 9900a422 Kuninori Morimoto 2017-09-25 3676 if (source->driver->ops->startup) { 4a75aae1 Charles Keepax 2018-09-05 3677 ret = source->driver->ops->startup(&substream, 4a75aae1 Charles Keepax 2018-09-05 3678 source); 9b8ef9f6 Jeeja KP 2015-10-20 3679 if (ret < 0) { 9b8ef9f6 Jeeja KP 2015-10-20 3680 dev_err(source->dev, 4a75aae1 Charles Keepax 2018-09-05 3681 "ASoC: startup() failed: %d\n", 4a75aae1 Charles Keepax 2018-09-05 3682 ret); 9b8ef9f6 Jeeja KP 2015-10-20 3683 goto out; 9b8ef9f6 Jeeja KP 2015-10-20 3684 } 9b8ef9f6 Jeeja KP 2015-10-20 3685 source->active++; 9b8ef9f6 Jeeja KP 2015-10-20 3686 } 93e6958a Benoit Cousson 2014-07-08 3687 ret = soc_dai_hw_params(&substream, params, source); 93e6958a Benoit Cousson 2014-07-08 3688 if (ret < 0) 9747cec2 Mark Brown 2012-04-26 3689 goto out; 4a75aae1 Charles Keepax 2018-09-05 3690 } c74184ed Mark Brown 2012-04-04 3691 c74184ed Mark Brown 2012-04-04 3692 substream.stream = SNDRV_PCM_STREAM_PLAYBACK; 4a75aae1 Charles Keepax 2018-09-05 3693 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3694 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3695 9900a422 Kuninori Morimoto 2017-09-25 3696 if (sink->driver->ops->startup) { 4a75aae1 Charles Keepax 2018-09-05 3697 ret = sink->driver->ops->startup(&substream, 4a75aae1 Charles Keepax 2018-09-05 3698 sink); 9b8ef9f6 Jeeja KP 2015-10-20 3699 if (ret < 0) { 9b8ef9f6 Jeeja KP 2015-10-20 3700 dev_err(sink->dev, 4a75aae1 Charles Keepax 2018-09-05 3701 "ASoC: startup() failed: %d\n", 4a75aae1 Charles Keepax 2018-09-05 3702 ret); 9b8ef9f6 Jeeja KP 2015-10-20 3703 goto out; 9b8ef9f6 Jeeja KP 2015-10-20 3704 } 9b8ef9f6 Jeeja KP 2015-10-20 3705 sink->active++; 9b8ef9f6 Jeeja KP 2015-10-20 3706 } 93e6958a Benoit Cousson 2014-07-08 3707 ret = soc_dai_hw_params(&substream, params, sink); 93e6958a Benoit Cousson 2014-07-08 3708 if (ret < 0) 9747cec2 Mark Brown 2012-04-26 3709 goto out; 4a75aae1 Charles Keepax 2018-09-05 3710 } c74184ed Mark Brown 2012-04-04 3711 break; c74184ed Mark Brown 2012-04-04 3712 c74184ed Mark Brown 2012-04-04 3713 case SND_SOC_DAPM_POST_PMU: 4a75aae1 Charles Keepax 2018-09-05 3714 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3715 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3716 da18396f Mark Brown 2013-02-06 3717 ret = snd_soc_dai_digital_mute(sink, 0, da18396f Mark Brown 2013-02-06 3718 SNDRV_PCM_STREAM_PLAYBACK); c74184ed Mark Brown 2012-04-04 3719 if (ret != 0 && ret != -ENOTSUPP) 4a75aae1 Charles Keepax 2018-09-05 3720 dev_warn(sink->dev, 4a75aae1 Charles Keepax 2018-09-05 3721 "ASoC: Failed to unmute: %d\n", ret); 9747cec2 Mark Brown 2012-04-26 3722 ret = 0; 4a75aae1 Charles Keepax 2018-09-05 3723 } c74184ed Mark Brown 2012-04-04 3724 break; c74184ed Mark Brown 2012-04-04 3725 c74184ed Mark Brown 2012-04-04 3726 case SND_SOC_DAPM_PRE_PMD: 4a75aae1 Charles Keepax 2018-09-05 3727 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3728 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3729 da18396f Mark Brown 2013-02-06 3730 ret = snd_soc_dai_digital_mute(sink, 1, da18396f Mark Brown 2013-02-06 3731 SNDRV_PCM_STREAM_PLAYBACK); c74184ed Mark Brown 2012-04-04 3732 if (ret != 0 && ret != -ENOTSUPP) 4a75aae1 Charles Keepax 2018-09-05 3733 dev_warn(sink->dev, 4a75aae1 Charles Keepax 2018-09-05 3734 "ASoC: Failed to mute: %d\n", ret); 9747cec2 Mark Brown 2012-04-26 3735 ret = 0; 4a75aae1 Charles Keepax 2018-09-05 3736 } 4a75aae1 Charles Keepax 2018-09-05 3737 4a75aae1 Charles Keepax 2018-09-05 3738 snd_soc_dapm_widget_for_each_source_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3739 source = path->source->priv; 9b8ef9f6 Jeeja KP 2015-10-20 3740 9b8ef9f6 Jeeja KP 2015-10-20 3741 source->active--; 9900a422 Kuninori Morimoto 2017-09-25 3742 if (source->driver->ops->shutdown) { 9b8ef9f6 Jeeja KP 2015-10-20 3743 substream.stream = SNDRV_PCM_STREAM_CAPTURE; 4a75aae1 Charles Keepax 2018-09-05 3744 source->driver->ops->shutdown(&substream, 4a75aae1 Charles Keepax 2018-09-05 3745 source); 4a75aae1 Charles Keepax 2018-09-05 3746 } 9b8ef9f6 Jeeja KP 2015-10-20 3747 } 9b8ef9f6 Jeeja KP 2015-10-20 3748 4a75aae1 Charles Keepax 2018-09-05 3749 snd_soc_dapm_widget_for_each_sink_path(w, path) { 4a75aae1 Charles Keepax 2018-09-05 3750 sink = path->sink->priv; 4a75aae1 Charles Keepax 2018-09-05 3751 9b8ef9f6 Jeeja KP 2015-10-20 3752 sink->active--; 9900a422 Kuninori Morimoto 2017-09-25 3753 if (sink->driver->ops->shutdown) { 9b8ef9f6 Jeeja KP 2015-10-20 3754 substream.stream = SNDRV_PCM_STREAM_PLAYBACK; 9b8ef9f6 Jeeja KP 2015-10-20 3755 sink->driver->ops->shutdown(&substream, sink); 9b8ef9f6 Jeeja KP 2015-10-20 3756 } 4a75aae1 Charles Keepax 2018-09-05 3757 } c74184ed Mark Brown 2012-04-04 3758 break; c74184ed Mark Brown 2012-04-04 3759 c74184ed Mark Brown 2012-04-04 3760 default: a6ed0608 Takashi Iwai 2013-11-06 3761 WARN(1, "Unknown event %d\n", event); 75881df3 Sudip Mukherjee 2015-09-10 3762 ret = -EINVAL; c74184ed Mark Brown 2012-04-04 3763 } c74184ed Mark Brown 2012-04-04 3764 9747cec2 Mark Brown 2012-04-26 3765 out: 8053f216 Nicolin Chen 2016-07-26 3766 kfree(runtime); 9747cec2 Mark Brown 2012-04-26 3767 kfree(params); 9747cec2 Mark Brown 2012-04-26 @3768 return ret; c74184ed Mark Brown 2012-04-04 3769 } c74184ed Mark Brown 2012-04-04 3770 :::::: The code at line 3768 was first introduced by commit :::::: 9747cec21e2d55ee996249b01ec30dd32389c0ba ASoC: dapm: Move CODEC<->CODEC params off stack :::::: TO: Mark Brown <broonie(a)opensource.wolfsonmicro.com> :::::: CC: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: rsnd: Add device tree binding for r8a77990
by Yoshihiro Kaneko 06 Sep '18

06 Sep '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> This patch adds the device tree binding of the r8a77990 SoC. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> --- This patch is based on the devel branch of Simon Horman's renesas tree. Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790..07aba8d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77990" (R-Car E3) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 1.9.1

5 5

[alsa-devel] [PATCH] ASoC: sgtl5000: avoid division by zero if lo_vag is zero
by Colin King 06 Sep '18

06 Sep '18

From: Colin Ian King <colin.king(a)canonical.com> In the case where lo_vag <= SGTL5000_LINE_OUT_GND_BASE, lo_vag is set to zero and later vol_quot is computed by dividing by lo_vag causing a division by zero error. Fix this by avoiding a zero division and set vol_quot to zero in this specific case so that the lowest setting for i is correctly set. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 60764f6201b1..add18d6d77da 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1218,7 +1218,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_component *component) * Searching for a suitable index solving this formula: * idx = 40 * log10(vag_val / lo_cagcntrl) + 15 */ - vol_quot = (vag * 100) / lo_vag; + vol_quot = lo_vag ? (vag * 100) / lo_vag : 0; lo_vol = 0; for (i = 0; i < ARRAY_SIZE(vol_quot_table); i++) { if (vol_quot >= vol_quot_table[i]) -- 2.17.1

4 3

[alsa-devel] Applied "ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3bbf5d34fd4a0c41246290b70338095ae291851b Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:20:58 +0100 Subject: [PATCH] ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked Currently DAPM has a lot of similar code to handle errors from snd_soc_dapm_new_control_unlocked, and much of this code does not really accurately reflect what the function returns. Firstly, most places will check for a return value of -EPROBE_DEFER and silence any error messages in that case. The one notable exception here being dapm_kcontrol_data_alloc which does currently print any error messages in the case of snd_soc_dapm_new_control_unlocked returning NULL or an error. Additionally the error prints being silenced in these case are redundant as snd_soc_dapm_new_control_unlocked can only return -EPROBE_DEFER or NULL when failing. Secondly, most places will treat a return value of NULL as an -ENOMEM. This is not correct either since any error except EPROBE_DEFER will cause a return value of NULL from snd_soc_dapm_new_control_unlocked. Centralise this handling and the error messages within snd_soc_dapm_new_control_unlocked and update the callers to simply check IS_ERR and return. Note that this update is slightly simpler in the case of dapm_kcontrol_data_alloc where that is fairly close to the handling that was already in place. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 118 +++++++-------------------------------- sound/soc/soc-topology.c | 11 ---- 2 files changed, 19 insertions(+), 110 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a738cb439be..d13a25ce1275 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -364,10 +364,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } } break; case snd_soc_dapm_demux: @@ -402,10 +398,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); @@ -3433,23 +3425,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); - /* Do not nag about probe deferrals */ - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - goto out_unlock; - } - if (!w) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - -out_unlock: mutex_unlock(&dapm->card->dapm_mutex); + return w; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); @@ -3464,24 +3441,20 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return NULL; + return ERR_PTR(-ENOMEM); switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) - dev_warn(w->dapm->dev, + dev_warn(dapm->dev, "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3490,22 +3463,14 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->pinctrl = devm_pinctrl_get(dapm->dev); if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; default: @@ -3519,7 +3484,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return NULL; + return ERR_PTR(-ENOMEM); } switch (w->id) { @@ -3596,6 +3561,13 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; + +request_failed: + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + + return ERR_PTR(ret); } /** @@ -3621,19 +3593,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret == -EPROBE_DEFER) - break; - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - break; - } - if (!w) { - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - ret = -ENOMEM; break; } widget++; @@ -3944,21 +3903,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) { - ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(card->dev, - "ASoC: Failed to create %s widget (%d)\n", - link_name, ret); - goto outfree_kcontrol_news; - } - if (!w) { - dev_err(card->dev, "ASoC: Failed to create %s widget\n", - link_name); - ret = -ENOMEM; + if (IS_ERR(w)) goto outfree_kcontrol_news; - } w->params = params; w->num_params = num_params; @@ -3999,21 +3945,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->playback.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->playback_widget = w; @@ -4028,21 +3961,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->capture.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->capture_widget = w; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..17f81b9a5754 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1565,17 +1565,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, widget = snd_soc_dapm_new_control_unlocked(dapm, &template); if (IS_ERR(widget)) { ret = PTR_ERR(widget); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(tplg->dev, - "ASoC: failed to create widget %s controls (%d)\n", - w->name, ret); - goto hdr_err; - } - if (widget == NULL) { - dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", - w->name); - ret = -ENOMEM; goto hdr_err; } -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 94e630a35d3383b42f12a873a5404bdf61e38e42 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:20:59 +0100 Subject: [PATCH] ASoC: dapm: Cosmetic tidy up of snd_soc_dapm_new_control Move the function snd_soc_dapm_new_control to be next to snd_soc_dapm_new_controls and add some kernel doc for it. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 37 +++++++++++++++++++++++-------------- 1 file changed, 23 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d13a25ce1275..c111e69b9a09 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3417,20 +3417,6 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget) -{ - struct snd_soc_dapm_widget *w; - - mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - w = snd_soc_dapm_new_control_unlocked(dapm, widget); - mutex_unlock(&dapm->card->dapm_mutex); - - return w; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); - struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) @@ -3570,6 +3556,29 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, return ERR_PTR(ret); } +/** + * snd_soc_dapm_new_control - create new dapm control + * @dapm: DAPM context + * @widget: widget template + * + * Creates new DAPM control based upon a template. + * + * Returns a widget pointer on success or an error pointer on failure + */ +struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) +{ + struct snd_soc_dapm_widget *w; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + w = snd_soc_dapm_new_control_unlocked(dapm, widget); + mutex_unlock(&dapm->card->dapm_mutex); + + return w; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); + /** * snd_soc_dapm_new_controls - create new dapm controls * @dapm: DAPM context -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH v2 1/5] ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked
by Charles Keepax 06 Sep '18

06 Sep '18

Currently DAPM has a lot of similar code to handle errors from snd_soc_dapm_new_control_unlocked, and much of this code does not really accurately reflect what the function returns. Firstly, most places will check for a return value of -EPROBE_DEFER and silence any error messages in that case. The one notable exception here being dapm_kcontrol_data_alloc which does currently print any error messages in the case of snd_soc_dapm_new_control_unlocked returning NULL or an error. Additionally the error prints being silenced in these case are redundant as snd_soc_dapm_new_control_unlocked can only return -EPROBE_DEFER or NULL when failing. Secondly, most places will treat a return value of NULL as an -ENOMEM. This is not correct either since any error except EPROBE_DEFER will cause a return value of NULL from snd_soc_dapm_new_control_unlocked. Centralise this handling and the error messages within snd_soc_dapm_new_control_unlocked and update the callers to simply check IS_ERR and return. Note that this update is slightly simpler in the case of dapm_kcontrol_data_alloc where that is fairly close to the handling that was already in place. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Improved the commit message to be a more descriptive of the change Thanks, Charles sound/soc/soc-dapm.c | 118 ++++++++--------------------------------------- sound/soc/soc-topology.c | 11 ----- 2 files changed, 19 insertions(+), 110 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a738cb439be..d13a25ce1275 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -364,10 +364,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } } break; case snd_soc_dapm_demux: @@ -402,10 +398,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, ret = PTR_ERR(data->widget); goto err_data; } - if (!data->widget) { - ret = -ENOMEM; - goto err_data; - } snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); @@ -3433,23 +3425,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); - /* Do not nag about probe deferrals */ - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - goto out_unlock; - } - if (!w) - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - -out_unlock: mutex_unlock(&dapm->card->dapm_mutex); + return w; } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); @@ -3464,24 +3441,20 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return NULL; + return ERR_PTR(-ENOMEM); switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } if (w->on_val & SND_SOC_DAPM_REGULATOR_BYPASS) { ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) - dev_warn(w->dapm->dev, + dev_warn(dapm->dev, "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -3490,22 +3463,14 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->pinctrl = devm_pinctrl_get(dapm->dev); if (IS_ERR(w->pinctrl)) { ret = PTR_ERR(w->pinctrl); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; case snd_soc_dapm_clock_supply: w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); - if (ret == -EPROBE_DEFER) - return ERR_PTR(ret); - dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", - w->name, ret); - return NULL; + goto request_failed; } break; default: @@ -3519,7 +3484,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name = kstrdup_const(widget->name, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return NULL; + return ERR_PTR(-ENOMEM); } switch (w->id) { @@ -3596,6 +3561,13 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; + +request_failed: + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + + return ERR_PTR(ret); } /** @@ -3621,19 +3593,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, w = snd_soc_dapm_new_control_unlocked(dapm, widget); if (IS_ERR(w)) { ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret == -EPROBE_DEFER) - break; - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s (%d)\n", - widget->name, ret); - break; - } - if (!w) { - dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s\n", - widget->name); - ret = -ENOMEM; break; } widget++; @@ -3944,21 +3903,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) { - ret = PTR_ERR(w); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(card->dev, - "ASoC: Failed to create %s widget (%d)\n", - link_name, ret); - goto outfree_kcontrol_news; - } - if (!w) { - dev_err(card->dev, "ASoC: Failed to create %s widget\n", - link_name); - ret = -ENOMEM; + if (IS_ERR(w)) goto outfree_kcontrol_news; - } w->params = params; w->num_params = num_params; @@ -3999,21 +3945,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->playback.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->playback_widget = w; @@ -4028,21 +3961,8 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); - if (IS_ERR(w)) { - int ret = PTR_ERR(w); - - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(dapm->dev, - "ASoC: Failed to create %s widget (%d)\n", - dai->driver->playback.stream_name, ret); - return ret; - } - if (!w) { - dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", - dai->driver->capture.stream_name); - return -ENOMEM; - } + if (IS_ERR(w)) + return PTR_ERR(w); w->priv = dai; dai->capture_widget = w; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 66e77e020745..17f81b9a5754 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1565,17 +1565,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, widget = snd_soc_dapm_new_control_unlocked(dapm, &template); if (IS_ERR(widget)) { ret = PTR_ERR(widget); - /* Do not nag about probe deferrals */ - if (ret != -EPROBE_DEFER) - dev_err(tplg->dev, - "ASoC: failed to create widget %s controls (%d)\n", - w->name, ret); - goto hdr_err; - } - if (widget == NULL) { - dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", - w->name); - ret = -ENOMEM; goto hdr_err; } -- 2.11.0

2 5

[alsa-devel] Applied "ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4a75aae17b2a802a7267206414050408392c374c Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:21:01 +0100 Subject: [PATCH] ASoC: dapm: Add support for multi-CODEC CODEC to CODEC links Currently multi-CODEC is not supported on CODEC to CODEC links. There are common applications where this would be useful, such as connecting two mono amplifiers to an audio CODEC. Adding support simply requires an update of snd_soc_dai_link_event to loop over the attached CODEC DAIs. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++------------------- 1 file changed, 70 insertions(+), 53 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bbfcb7fc05cc..40f27c95da61 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3614,7 +3614,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_dapm_path *source_p, *sink_p; + struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; const struct snd_soc_pcm_stream *config = w->params + w->params_select; @@ -3629,17 +3629,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) return -EINVAL; - /* We only support a single source and sink, pick the first */ - source_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_OUT], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_OUT]); - sink_p = list_first_entry(&w->edges[SND_SOC_DAPM_DIR_IN], - struct snd_soc_dapm_path, - list_node[SND_SOC_DAPM_DIR_IN]); - - source = source_p->source->priv; - sink = sink_p->sink->priv; - /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3681,59 +3670,90 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; - if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; + + if (source->driver->ops->startup) { + ret = source->driver->ops->startup(&substream, + source); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + source->active++; } - source->active++; + ret = soc_dai_hw_params(&substream, params, source); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, source); - if (ret < 0) - goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", ret); - goto out; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + if (sink->driver->ops->startup) { + ret = sink->driver->ops->startup(&substream, + sink); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", + ret); + goto out; + } + sink->active++; } - sink->active++; + ret = soc_dai_hw_params(&substream, params, sink); + if (ret < 0) + goto out; } - ret = soc_dai_hw_params(&substream, params, sink); - if (ret < 0) - goto out; break; case SND_SOC_DAPM_POST_PMU: - ret = snd_soc_dai_digital_mute(sink, 0, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to unmute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 0, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to unmute: %d\n", ret); + ret = 0; + } break; case SND_SOC_DAPM_PRE_PMD: - ret = snd_soc_dai_digital_mute(sink, 1, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(sink->dev, "ASoC: Failed to mute: %d\n", ret); - ret = 0; + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + ret = snd_soc_dai_digital_mute(sink, 1, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret != 0 && ret != -ENOTSUPP) + dev_warn(sink->dev, + "ASoC: Failed to mute: %d\n", ret); + ret = 0; + } + + snd_soc_dapm_widget_for_each_source_path(w, path) { + source = path->source->priv; - source->active--; - if (source->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_CAPTURE; - source->driver->ops->shutdown(&substream, source); + source->active--; + if (source->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_CAPTURE; + source->driver->ops->shutdown(&substream, + source); + } } - sink->active--; - if (sink->driver->ops->shutdown) { - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - sink->driver->ops->shutdown(&substream, sink); + snd_soc_dapm_widget_for_each_sink_path(w, path) { + sink = path->sink->priv; + + sink->active--; + if (sink->driver->ops->shutdown) { + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + sink->driver->ops->shutdown(&substream, sink); + } } break; @@ -4043,9 +4063,6 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, int i; if (rtd->dai_link->params) { - if (rtd->num_codecs > 1) - dev_warn(card->dev, "ASoC: Multiple codecs not supported yet\n"); - playback_cpu = cpu_dai->capture_widget; capture_cpu = cpu_dai->playback_widget; } else { -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime" to the asoc tree
by Mark Brown 06 Sep '18

06 Sep '18

The patch ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 243bcfafcd9a23a20867fd488dc3a35264918d87 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Wed, 5 Sep 2018 15:21:02 +0100 Subject: [PATCH] ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime Larger CODECs may contain many several hundred widgets and which set of parameters is selected only needs to be recorded on a per DAI basis. As such move the selected CODEC to CODEC link params to be stored in the runtime rather than the DAPM widget, to save some memory. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dapm.h | 3 --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 19 +++++++++++-------- 3 files changed, 13 insertions(+), 11 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index cb177fa21ce7..bd8163f151cb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -584,9 +584,6 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ struct pinctrl *pinctrl; /* attached pinctrl */ - const struct snd_soc_pcm_stream *params; /* params for dai links */ - unsigned int num_params; /* number of params for dai links */ - unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6b68b31e3140..821bf99992b8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1130,6 +1130,8 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; + unsigned int params_select; /* currently selected param for dai link */ + /* Dynamic PCM BE runtime data */ struct snd_soc_dpcm_runtime dpcm[2]; int fe_compr; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 40f27c95da61..9f4edcd19e02 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1018,9 +1018,10 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_kcontrol *kcontrol; struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; + struct snd_soc_pcm_runtime *rtd = w->priv; /* create control for links with > 1 config */ - if (w->num_params <= 1) + if (rtd->dai_link->num_params <= 1) return 0; /* add kcontrol */ @@ -3617,13 +3618,15 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_path *path; struct snd_soc_dai *source, *sink; struct snd_soc_pcm_runtime *rtd = w->priv; - const struct snd_soc_pcm_stream *config = w->params + w->params_select; + const struct snd_soc_pcm_stream *config; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; int ret; + config = rtd->dai_link->params + rtd->params_select; + if (WARN_ON(!config) || WARN_ON(list_empty(&w->edges[SND_SOC_DAPM_DIR_OUT]) || list_empty(&w->edges[SND_SOC_DAPM_DIR_IN]))) @@ -3772,8 +3775,9 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; - ucontrol->value.enumerated.item[0] = w->params_select; + ucontrol->value.enumerated.item[0] = rtd->params_select; return 0; } @@ -3782,18 +3786,19 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_pcm_runtime *rtd = w->priv; /* Can't change the config when widget is already powered */ if (w->power) return -EBUSY; - if (ucontrol->value.enumerated.item[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == rtd->params_select) return 0; - if (ucontrol->value.enumerated.item[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= rtd->dai_link->num_params) return -EINVAL; - w->params_select = ucontrol->value.enumerated.item[0]; + rtd->params_select = ucontrol->value.enumerated.item[0]; return 0; } @@ -3936,8 +3941,6 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, if (IS_ERR(w)) goto outfree_kcontrol_news; - w->params = rtd->dai_link->params; - w->num_params = rtd->dai_link->num_params; w->priv = rtd; return w; -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH] ASoC: hdac_hdmi: remove redundant check for !port condition
by Colin King 06 Sep '18

06 Sep '18

From: Colin Ian King <colin.king(a)canonical.com> The !port check is redundant as it being performed in the following check. Remove it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdac_hdmi.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..dc6a0dfea050 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1961,9 +1961,6 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdev, int pcm_idx) port = list_first_entry(&pcm->port_list, struct hdac_hdmi_port, head); - if (!port) - return 0; - if (!port || !port->eld.eld_valid) return 0; -- 2.17.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: rsnd: bugfix patches for linus/master, and v4.20
by Kuninori Morimoto 06 Sep '18

06 Sep '18

Hi Mark These are bug fix patches for Renesas Sound driver. 1) : for mark/for-4.20 2) - 4) : for linus/master Kuninori Morimoto (4): 1) ASoC: rsnd: gen: use tab instead of white-space 2) ASoC: rsnd: adg: care clock-frequency size 3) ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER 4) ASoC: rsnd: don't use %p for dev_dbg() sound/soc/sh/rcar/adg.c | 9 +++++++-- sound/soc/sh/rcar/core.c | 10 +++++++++- sound/soc/sh/rcar/dma.c | 6 +++++- sound/soc/sh/rcar/gen.c | 48 ++++++++++++++++++++++++------------------------ 4 files changed, 45 insertions(+), 28 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 8

[alsa-devel] [PATCH 1/1] ASoC: cs4265: fix MMTLR Data switch control
by Sébastien Szymanski 06 Sep '18

06 Sep '18

The MMTLR bit is in the CS4265_SPDIF_CTL2 register at address 0x12 bit 0 and not at address 0x0 bit 1. Fix this. Signed-off-by: Sébastien Szymanski <sebastien.szymanski(a)armadeus.com> --- sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de669f..407554175282 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,8 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, + 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.16.4

1 0

[alsa-devel] [PATCH 0/1] Fix ADI axi-i2s + Xilinx AXI-DMA capture
by Luca Ceresoli 06 Sep '18

06 Sep '18

Hi, here is a fix for a nasty audio capture problem when the axi-i2s output stream is fed to a Xilinx AXI-DMA. The commit is simple and hopefully well described, but I am not 100% sure the solution is the correct one. If it is not, I'll be glad to know which one is the best. It solved my problem anyway. Luca Luca Ceresoli (1): axi-i2s: set period size register sound/soc/adi/axi-i2s.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) -- 2.17.1

4 8

[alsa-devel] [RFC] UAPI: Check headers by compiling all together as C++
by David Howells 06 Sep '18

06 Sep '18

Here's a set of patches that inserts a step into the build process to make sure that the UAPI headers can all be built together with C++ (if the compiler being used supports C++). All but the final patch perform fixups, including: (1) Fix member names that conflict with C++ reserved words by providing alternates that can be used anywhere. An anonymous union is used so that that the conflicting name is still available outside of C++. (2) Fix the use of flexible arrays in structs that get embedded (which is illegal in C++). (3) Remove the use of internal kernel structs in UAPI structures. (4) Fix symbol collisions. (5) Replace usage of u32 and co. with __u32 and co. (6) Fix use of sparsely initialised arrays (which g++ doesn't implement). (7) Remove some use of PAGE_SIZE since this isn't valid outside of the kernel. And lastly: (8) Compile all of the UAPI headers (with a few exceptions) together as C++ to catch new errors occurring as part of the regular build process. The patches can also be found here: http://git.kernel.org/cgit/linux/kernel/git/dhowells/linux-fs.git/log/?h=ua… Thanks, David --- David Howells (11): UAPI: drm: Fix use of C++ keywords as structural members UAPI: keys: Fix use of C++ keywords as structural members UAPI: virtio_net: Fix use of C++ keywords as structural members UAPI: bcache: Fix use of embedded flexible array UAPI: coda: Don't use internal kernel structs in UAPI UAPI: netfilter: Fix symbol collision issues UAPI: nilfs2: Fix use of undefined byteswapping functions UAPI: sound: Fix use of u32 and co. in UAPI headers UAPI: ndctl: Fix g++-unsupported initialisation in headers UAPI: ndctl: Remove use of PAGE_SIZE UAPI: Check headers build for C++ Makefile | 1 include/linux/ndctl.h | 22 ++++ include/uapi/drm/i810_drm.h | 7 + include/uapi/drm/msm_drm.h | 7 + include/uapi/linux/bcache.h | 2 include/uapi/linux/coda_psdev.h | 4 + include/uapi/linux/keyctl.h | 7 + include/uapi/linux/ndctl.h | 20 ++- include/uapi/linux/netfilter/nfnetlink_cthelper.h | 2 include/uapi/linux/netfilter_ipv4/ipt_ECN.h | 9 -- include/uapi/linux/nilfs2_ondisk.h | 21 ++-- include/uapi/linux/virtio_net.h | 7 + include/uapi/sound/skl-tplg-interface.h | 106 +++++++++--------- scripts/headers-c++.sh | 124 +++++++++++++++++++++ 14 files changed, 255 insertions(+), 84 deletions(-) create mode 100644 include/linux/ndctl.h create mode 100755 scripts/headers-c++.sh

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[alsa-devel] [PATCH] ASoC: uniphier: change status to orphan
by Katsuhiro Suzuki 05 Sep '18

05 Sep '18

Since I'm leaving from Socionext, I'll unable to access specification documents of this hardware (these are not public). So change the state to orphan until someone will maintain this driver. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- MAINTAINERS | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a8683e549777..4788a5335d35 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13450,9 +13450,8 @@ F: drivers/i2c/busses/i2c-synquacer.c F: Documentation/devicetree/bindings/i2c/i2c-synquacer.txt SOCIONEXT UNIPHIER SOUND DRIVER -M: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Orphan F: sound/soc/uniphier/ SOEKRIS NET48XX LED SUPPORT -- 2.18.0

2 2

[alsa-devel] [PATCH v2 1/3] ASoC: tlv320aic31xx: Add overflow detection support
by Andrew F. Davis 04 Sep '18

04 Sep '18

Similar to short circuit detection, when the ADC/DAC is saturated and overflows poor audio quality can result and should be reported to the user. This device support Automatic Dynamic Range Compression (DRC) to reduce this but it is not enabled currently in this driver. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Change errors to warning for non-fatal events - Rebased on broonie/sound.git for-next sound/soc/codecs/tlv320aic31xx.c | 34 ++++++++++++++++++++++++++++++-- sound/soc/codecs/tlv320aic31xx.h | 7 +++++++ 2 files changed, 39 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d8783af7d31c..6e8f79ec9629 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1356,7 +1356,7 @@ static irqreturn_t aic31xx_irq(int irq, void *data) if (value) handled = true; else - goto exit; + goto read_overflow; if (value & AIC31XX_HPLSCDETECT) dev_err(dev, "Short circuit on Left output is detected\n"); @@ -1366,6 +1366,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) AIC31XX_HPRSCDETECT)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); +read_overflow: + ret = regmap_read(aic31xx->regmap, AIC31XX_OFFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read overflow flag: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_DAC_OF_LEFT) + dev_warn(dev, "Left-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_RIGHT) + dev_warn(dev, "Right-channel DAC overflow has occurred\n"); + if (value & AIC31XX_DAC_OF_SHIFTER) + dev_warn(dev, "DAC barrel shifter overflow has occurred\n"); + if (value & AIC31XX_ADC_OF) + dev_warn(dev, "ADC overflow has occurred\n"); + if (value & AIC31XX_ADC_OF_SHIFTER) + dev_warn(dev, "ADC barrel shifter overflow has occurred\n"); + if (value & ~(AIC31XX_DAC_OF_LEFT | + AIC31XX_DAC_OF_RIGHT | + AIC31XX_DAC_OF_SHIFTER | + AIC31XX_ADC_OF | + AIC31XX_ADC_OF_SHIFTER)) + dev_warn(dev, "Unknown overflow interrupt flags: 0x%08x\n", value); + exit: if (handled) return IRQ_HANDLED; @@ -1447,7 +1476,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, - AIC31XX_SC); + AIC31XX_SC | + AIC31XX_ENGINE); ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, NULL, aic31xx_irq, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52e171988906..2636f2c6bc79 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -173,6 +173,13 @@ struct aic31xx_pdata { #define AIC31XX_HPRDRVPWRSTATUS_MASK BIT(1) #define AIC31XX_SPRDRVPWRSTATUS_MASK BIT(0) +/* AIC31XX_OFFLAG */ +#define AIC31XX_DAC_OF_LEFT BIT(7) +#define AIC31XX_DAC_OF_RIGHT BIT(6) +#define AIC31XX_DAC_OF_SHIFTER BIT(5) +#define AIC31XX_ADC_OF BIT(3) +#define AIC31XX_ADC_OF_SHIFTER BIT(1) + /* AIC31XX_INTRDACFLAG */ #define AIC31XX_HPLSCDETECT BIT(7) #define AIC31XX_HPRSCDETECT BIT(6) -- 2.18.0

2 4

[alsa-devel] [PATCH 0/2][RFC] ASoC: add for_each_xxx_codec macro
by Kuninori Morimoto 04 Sep '18

04 Sep '18

Hi Mark These are not so important patches. But current code is using for loop for dai_link codecs and rtd codec_dai, but I think using for_each_xxx macro for these can be more readable. These patches do it. Kuninori Morimoto (2): ASoC: add for_each_link_codecs() macro ASoC: add for_each_rtd_codec_dai() macro include/sound/soc.h | 11 ++ sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 5 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- sound/soc/meson/axg-card.c | 11 +- sound/soc/soc-core.c | 55 ++++---- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-pcm.c | 154 ++++++++++----------- 10 files changed, 133 insertions(+), 137 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Add MICBIAS off setting
by Andrew F. Davis 04 Sep '18

04 Sep '18

Leaving microphone bias off is a valid setting and even used in the DT binding document example. Add this setting here and document the same. Although it may not make much sense to enable a microphone here without any bias, it is a valid setting that can be chosen by DT and may be needed for some boards. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 1 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 1 + sound/soc/codecs/tlv320aic31xx.c | 1 + 3 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt index 5b3c33bb99e5..411cc46a2c58 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -24,6 +24,7 @@ Optional properties: - reset-gpios - GPIO specification for the active low RESET input. - ai31xx-micbias-vg - MicBias Voltage setting + 0 or MICBIAS_OFF - MICBIAS output is powered off 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h index c6895a18a455..069484070fcf 100644 --- a/include/dt-bindings/sound/tlv320aic31xx-micbias.h +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -2,6 +2,7 @@ #ifndef __DT_TLV320AIC31XX_MICBIAS_H #define __DT_TLV320AIC31XX_MICBIAS_H +#define MICBIAS_OFF 0 #define MICBIAS_2_0V 1 #define MICBIAS_2_5V 2 #define MICBIAS_AVDDV 3 diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index bf92d36b8f8a..7d87df518fed 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1421,6 +1421,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, fwnode_property_read_u32(aic31xx->dev->fwnode, "ai31xx-micbias-vg", &micbias_value); switch (micbias_value) { + case MICBIAS_OFF: case MICBIAS_2_0V: case MICBIAS_2_5V: case MICBIAS_AVDDV: -- 2.18.0

2 9

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Remove regulator notification handling
by Andrew F. Davis 04 Sep '18

04 Sep '18

The notification handler sets the reset line but never re-enables it as no one is listening for an enable event, this is certainly broken and was most likely just copied from other CODECs, remove this code. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 55 +------------------------------- 1 file changed, 1 insertion(+), 54 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7d87df518fed..f8d0f3ebed9c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -147,11 +147,6 @@ static const char * const aic31xx_supply_names[] = { #define AIC31XX_NUM_SUPPLIES ARRAY_SIZE(aic31xx_supply_names) -struct aic31xx_disable_nb { - struct notifier_block nb; - struct aic31xx_priv *aic31xx; -}; - struct aic31xx_priv { struct snd_soc_component *component; u8 i2c_regs_status; @@ -162,7 +157,6 @@ struct aic31xx_priv { int micbias_vg; struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; - struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1116,28 +1110,6 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static int aic31xx_regulator_event(struct notifier_block *nb, - unsigned long event, void *data) -{ - struct aic31xx_disable_nb *disable_nb = - container_of(nb, struct aic31xx_disable_nb, nb); - struct aic31xx_priv *aic31xx = disable_nb->aic31xx; - - if (event & REGULATOR_EVENT_DISABLE) { - /* - * Put codec to reset and as at least one of the - * supplies was disabled. - */ - if (aic31xx->gpio_reset) - gpiod_set_value(aic31xx->gpio_reset, 1); - - regcache_mark_dirty(aic31xx->regmap); - dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); - } - - return 0; -} - static int aic31xx_reset(struct aic31xx_priv *aic31xx) { int ret = 0; @@ -1263,26 +1235,12 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i, ret; + int ret; dev_dbg(aic31xx->dev, "## %s\n", __func__); aic31xx->component = component; - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { - aic31xx->disable_nb[i].nb.notifier_call = - aic31xx_regulator_event; - aic31xx->disable_nb[i].aic31xx = aic31xx; - ret = regulator_register_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); - if (ret) { - dev_err(component->dev, - "Failed to request regulator notifier: %d\n", - ret); - return ret; - } - } - regcache_cache_only(aic31xx->regmap, true); regcache_mark_dirty(aic31xx->regmap); @@ -1297,19 +1255,8 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) return 0; } -static void aic31xx_codec_remove(struct snd_soc_component *component) -{ - struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); - int i; - - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) - regulator_unregister_notifier(aic31xx->supplies[i].consumer, - &aic31xx->disable_nb[i].nb); -} - static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, - .remove = aic31xx_codec_remove, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), -- 2.18.0

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[alsa-devel] [PATCH 1/4] ASoC: tlv320aic31xx: Add short circuit detection support
by Andrew F. Davis 04 Sep '18

04 Sep '18

These devices support detecting and reporting short circuits across the output stages. Add support for reporting these issue. Do this by registering an interrupt if available and enabling this error to trigger that interrupt in the device. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 16 ++++++++++ 2 files changed, 71 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8d0f3ebed9c..d8783af7d31c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -161,6 +161,7 @@ struct aic31xx_priv { u8 p_div; int rate_div_line; bool master_dapm_route_applied; + int irq; }; struct aic31xx_rate_divs { @@ -1338,6 +1339,40 @@ static const struct acpi_device_id aic31xx_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); #endif +static irqreturn_t aic31xx_irq(int irq, void *data) +{ + struct aic31xx_priv *aic31xx = data; + struct device *dev = aic31xx->dev; + unsigned int value; + bool handled = false; + int ret; + + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG, &value); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", ret); + goto exit; + } + + if (value) + handled = true; + else + goto exit; + + if (value & AIC31XX_HPLSCDETECT) + dev_err(dev, "Short circuit on Left output is detected\n"); + if (value & AIC31XX_HPRSCDETECT) + dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & ~(AIC31XX_HPLSCDETECT | + AIC31XX_HPRSCDETECT)) + dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); + +exit: + if (handled) + return IRQ_HANDLED; + else + return IRQ_NONE; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1360,6 +1395,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } aic31xx->dev = &i2c->dev; + aic31xx->irq = i2c->irq; aic31xx->codec_type = id->driver_data; @@ -1404,6 +1440,25 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return ret; } + if (aic31xx->irq > 0) { + regmap_update_bits(aic31xx->regmap, AIC31XX_GPIO1, + AIC31XX_GPIO1_FUNC_MASK, + AIC31XX_GPIO1_INT1 << + AIC31XX_GPIO1_FUNC_SHIFT); + + regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_SC); + + ret = devm_request_threaded_irq(aic31xx->dev, aic31xx->irq, + NULL, aic31xx_irq, + IRQF_ONESHOT, "aic31xx-irq", + aic31xx); + if (ret) { + dev_err(aic31xx->dev, "Unable to request IRQ\n"); + return ret; + } + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..52e171988906 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -191,6 +191,22 @@ struct aic31xx_pdata { #define AIC31XX_SC BIT(3) #define AIC31XX_ENGINE BIT(2) +/* AIC31XX_GPIO1 */ +#define AIC31XX_GPIO1_FUNC_MASK GENMASK(5, 2) +#define AIC31XX_GPIO1_FUNC_SHIFT 2 +#define AIC31XX_GPIO1_DISABLED 0x00 +#define AIC31XX_GPIO1_INPUT 0x01 +#define AIC31XX_GPIO1_GPI 0x02 +#define AIC31XX_GPIO1_GPO 0x03 +#define AIC31XX_GPIO1_CLKOUT 0x04 +#define AIC31XX_GPIO1_INT1 0x05 +#define AIC31XX_GPIO1_INT2 0x06 +#define AIC31XX_GPIO1_ADC_WCLK 0x07 +#define AIC31XX_GPIO1_SBCLK 0x08 +#define AIC31XX_GPIO1_SWCLK 0x09 +#define AIC31XX_GPIO1_ADC_MOD_CLK 0x10 +#define AIC31XX_GPIO1_SDOUT 0x11 + /* AIC31XX_DACSETUP */ #define AIC31XX_SOFTSTEP_MASK GENMASK(1, 0) -- 2.18.0

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[alsa-devel] [PATCH 0/5] Set SNDRV_PCM_INFO_SYNC_APPLPTR to indirect PCM usages
by Takashi Iwai 04 Sep '18

04 Sep '18

Hi, this is a patch set to pass the missing SNDRV_PCM_INFO_SYNC_APPLPTR flag to the drivers using the indirect PCM helpers. Also, one cleanup of the indirect PCM helper usage in mips/hal2 driver is included. Takashi === Takashi Iwai (5): ALSA: rme32: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: mips: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: emu10k1: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: cs46xx: Use SNDRV_PCM_INFO_SYNC_APPLPTR info flag ALSA: mips: Cleanup indirect PCM helper usages sound/mips/hal2.c | 13 +++++-------- sound/pci/cs46xx/cs46xx_lib.c | 6 ++++-- sound/pci/emu10k1/emupcm.c | 3 ++- sound/pci/rme32.c | 22 ++++++++-------------- 4 files changed, 19 insertions(+), 25 deletions(-) -- 2.18.0

2 6

[alsa-devel] Possibility of routing at the kernel level
by Mark Farrugia 04 Sep '18

04 Sep '18

Hi all, I am currently using the ALSA loopback driver to create a "virtual" sound card to abstract underlying real sound cards. I capture from the loopback in user-space, and then subsequently playback to a real sound card. The issue with this is that there is a significant latency introduced (~100ms) due to the number of transitions of the stream from user to kernel space. I am seeking a way to avoid this. My question is this: is there any way of routing PCM substreams at the kernel level, so that we can avoid the need to pipe audio back up to user-space with the loopback capturer? If there is no current solution that would achieve this, I would proposition to design and implement a system that uses a top-level ALSA driver, which then routes dynamically to one or many subscribed lower-level "audio path" drivers. The system would also have the option to loopback upon itself. Does anyone see any reason that I couldn't (or shouldn't) attempt such a solution? All feedback is welcome. Regards, Mark

2 3

[alsa-devel] [PATCH 00/13] ASoC: add full BUSIF support
by Kuninori Morimoto 04 Sep '18

04 Sep '18

Hi Mark Cc: Jiada, Timo, Dragos Dragos, Jiada, Timo added full BUSIF support, expand TDM mode, TDM split mode and its related features to Renesas sound driver on v4.14 base kernel. Thank you Dragos, Jiada, Timo. Unfortunately, I can't confirm expand TDM mode, TDM split mode on my environment. But I could confirm/test full BUSIF support. These patches are for it. Dragos Tarcatu (3): ASoC: rsnd: add support for 16/24 bit slot widths ASoC: rsnd: add support for the DSP_A/DSP_B formats ASoC: rsnd: add support for 8 bit S8 format Jiada Wang (9): ASoC: rsnd: add warning message to rsnd_kctrl_accept_runtime() ASoC: rsnd: remove is_play parameter from hw_rule function ASoC: rsnd: ssi: Fix issue in dma data address assignment ASoC: rsnd: ssi: Check runtime channel number rather than hw_params ASoc: rsnd: dma: Calculate dma address with consider of BUSIF ASoc: rsnd: dma: Calculate PDMACHCRE with consider of BUSIF ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0 ASoC: rsnd: ssiu: Support to init different BUSIF instance ASoC: rsnd: fixup not to call clk_get/set under non-atomic Kuninori Morimoto (1): ASoC: rsnd: merge .nolock_start and .prepare sound/soc/sh/rcar/core.c | 135 +++++++++++++++++++++++++---------------------- sound/soc/sh/rcar/dma.c | 100 ++++++++++++++++++++++------------- sound/soc/sh/rcar/gen.c | 27 ++++++++-- sound/soc/sh/rcar/rsnd.h | 64 ++++++++++++++++------ sound/soc/sh/rcar/ssi.c | 105 ++++++++++++++++++++++-------------- sound/soc/sh/rcar/ssiu.c | 92 +++++++++++++++++++++++++++----- 6 files changed, 349 insertions(+), 174 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: HDA: Fix several mismatch for register mask and value
by Keyon Jie 04 Sep '18

04 Sep '18

E.g. for snd_hdac_ext_link_clear_stream_id(), we should set (1 << stream) as mask, and 0 as value, here correct it and several similar mismatches. And, here also remove unreadable register_mask usage for those mask value updating. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/ext/hdac_ext_stream.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 1bd27576db98..a835558ddbc9 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -146,7 +146,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) { - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, 0, AZX_PPLCCTL_RUN); + snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, + AZX_PPLCCTL_RUN, AZX_PPLCCTL_RUN); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_start); @@ -171,7 +172,8 @@ void snd_hdac_ext_link_stream_reset(struct hdac_ext_stream *stream) snd_hdac_ext_link_stream_clear(stream); - snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, 0, AZX_PPLCCTL_STRST); + snd_hdac_updatel(stream->pplc_addr, AZX_REG_PPLCCTL, + AZX_PPLCCTL_STRST, AZX_PPLCCTL_STRST); udelay(3); timeout = 50; do { @@ -242,7 +244,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id); void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int stream) { - snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, 0, (1 << stream)); + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_link_clear_stream_id); @@ -415,7 +417,6 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; - u32 register_mask = 0; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); @@ -424,12 +425,8 @@ void snd_hdac_ext_stream_spbcap_enable(struct hdac_bus *bus, mask |= (1 << index); - register_mask = readl(bus->spbcap + AZX_REG_SPB_SPBFCCTL); - - mask |= register_mask; - if (enable) - snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, 0, mask); + snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, mask); else snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, 0); } @@ -503,7 +500,6 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; - u32 register_mask = 0; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); @@ -512,12 +508,8 @@ void snd_hdac_ext_stream_drsm_enable(struct hdac_bus *bus, mask |= (1 << index); - register_mask = readl(bus->drsmcap + AZX_REG_SPB_SPBFCCTL); - - mask |= register_mask; - if (enable) - snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, 0, mask); + snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, mask); else snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, 0); } -- 2.14.1

2 1

[alsa-devel] [PATCH v2] sound: Convert to using %pOFn instead of device_node.name
by Rob Herring 04 Sep '18

04 Sep '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Fix error in strncmp sound/aoa/soundbus/i2sbus/core.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 000b58522106..bd7c5029fc59 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child = NULL, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - static const char *rnames[] = { "i2sbus: %s (control)", - "i2sbus: %s (tx)", - "i2sbus: %s (rx)" }; + char node_name[6]; + static const char *rnames[] = { "i2sbus: %pOFn (control)", + "i2sbus: %pOFn (tx)", + "i2sbus: %pOFn (rx)" }; static irq_handler_t ints[] = { i2sbus_bus_intr, i2sbus_tx_intr, i2sbus_rx_intr }; - if (strlen(np->name) != 5) + if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5) return 0; - if (strncmp(np->name, "i2s-", 4)) + if (strncmp(node_name, "i2s-", 4)) return 0; dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); @@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio, dev->sound.pcmid = -1; dev->macio = macio; dev->control = control; - dev->bus_number = np->name[4] - 'a'; + dev->bus_number = node_name[4] - 'a'; INIT_LIST_HEAD(&dev->sound.codec_list); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { dev->interrupts[i] = -1; snprintf(dev->rnames[i], sizeof(dev->rnames[i]), - rnames[i], np->name); + rnames[i], np); } for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { int irq = irq_of_parse_and_map(np, i); -- 2.17.1

2 1

[alsa-devel] KMSAN: uninit-value in snd_midi_event_encode_byte
by syzbot 04 Sep '18

04 Sep '18

Hello, syzbot found the following crash on: HEAD commit: 2dca2cbde67a kmsan: fix build warnings with CONFIG_KMSAN=n git tree: https://github.com/google/kmsan.git/master console output: https://syzkaller.appspot.com/x/log.txt?x=1660ef6a400000 kernel config: https://syzkaller.appspot.com/x/.config?x=820d6393634b55e3 dashboard link: https://syzkaller.appspot.com/bug?extid=194dffdb8b22fc5d207a compiler: clang version 8.0.0 (trunk 339414) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+194dffdb8b22fc5d207a(a)syzkaller.appspotmail.com ================================================================== BUG: KMSAN: uninit-value in snd_midi_event_encode_byte+0x569/0xff0 sound/core/seq/seq_midi_event.c:195 CPU: 0 PID: 1623 Comm: kworker/0:1H Not tainted 4.19.0-rc1+ #36 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_vmidi_output_work Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x14b/0x190 lib/dump_stack.c:113 kmsan_report+0x183/0x2b0 mm/kmsan/kmsan.c:956 __msan_warning+0x70/0xc0 mm/kmsan/kmsan_instr.c:645 snd_midi_event_encode_byte+0x569/0xff0 sound/core/seq/seq_midi_event.c:195 snd_vmidi_output_work+0x34e/0x5b0 sound/core/seq/seq_virmidi.c:161 process_one_work+0x1605/0x1f40 kernel/workqueue.c:2153 worker_thread+0x11a2/0x2590 kernel/workqueue.c:2296 kthread+0x465/0x4a0 kernel/kthread.c:247 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:416 Uninit was stored to memory at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:256 [inline] kmsan_save_stack mm/kmsan/kmsan.c:271 [inline] kmsan_internal_chain_origin+0x128/0x210 mm/kmsan/kmsan.c:573 __msan_chain_origin+0x69/0xc0 mm/kmsan/kmsan_instr.c:482 __snd_rawmidi_transmit_peek sound/core/rawmidi.c:1103 [inline] snd_rawmidi_transmit+0xa75/0xbf0 sound/core/rawmidi.c:1228 snd_vmidi_output_work+0x2ac/0x5b0 sound/core/seq/seq_virmidi.c:159 process_one_work+0x1605/0x1f40 kernel/workqueue.c:2153 worker_thread+0x11a2/0x2590 kernel/workqueue.c:2296 kthread+0x465/0x4a0 kernel/kthread.c:247 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:416 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:256 [inline] kmsan_internal_poison_shadow+0xb8/0x1b0 mm/kmsan/kmsan.c:181 kmsan_kmalloc+0x98/0x100 mm/kmsan/kmsan_hooks.c:91 __kmalloc_node+0x7bf/0x11c0 mm/slub.c:3828 kmalloc_node include/linux/slab.h:555 [inline] kvmalloc_node+0x19d/0x3e0 mm/util.c:423 kvmalloc include/linux/mm.h:577 [inline] snd_rawmidi_runtime_create sound/core/rawmidi.c:132 [inline] open_substream+0x3c8/0xaa0 sound/core/rawmidi.c:276 rawmidi_open_priv+0x347/0x1000 sound/core/rawmidi.c:327 snd_rawmidi_open+0x7d4/0x1120 sound/core/rawmidi.c:424 soundcore_open+0x9be/0xa60 sound/sound_core.c:597 chrdev_open+0xc26/0xdb0 fs/char_dev.c:417 do_dentry_open+0xc9d/0x1660 fs/open.c:771 vfs_open+0xaf/0xe0 fs/open.c:880 do_last fs/namei.c:3418 [inline] path_openat+0x17a6/0x67c0 fs/namei.c:3534 do_filp_open+0x259/0x610 fs/namei.c:3564 do_sys_open+0x630/0x940 fs/open.c:1063 __do_sys_open fs/open.c:1081 [inline] __se_sys_open fs/open.c:1076 [inline] __x64_sys_open+0xf2/0x130 fs/open.c:1076 do_syscall_64+0x15b/0x220 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x63/0xe7 ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

3 16

[alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Håvard 03 Sep '18

03 Sep '18

Have tried enabling all GPIO settings as well as tried every possible pin combination I could think of. This problem has persisted for almost 4-years with no fix or patch to speak of. Would be great if someone finally had a look at it! -Håvard

1 0

[alsa-devel] [PATCH] ASoC: core: Don't schedule DAPM work if already in target state
by Jon Hunter 03 Sep '18

03 Sep '18

When dapm_power_widgets() is called, the dapm_pre_sequence_async() and dapm_post_sequence_async() functions are scheduled for all DAPM contexts (apart from the card DAPM context) regardless of whether the DAPM context is already in the desired state. The overhead of this is not insignificant and the more DAPM contexts there are the more overhead there is. For example, on the Tegra124 Jetson TK1, when profiling the time taken to execute the dapm_power_widgets() the following times were observed. Times for function dapm_power_widgets() are (us): Min 23, Ave 190, Max 434, Count 39 Here 'Count' is the number of times that dapm_power_widgets() has been called. Please note that the above time were measured using ktime_get() to log the time on entry and exit from dapm_power_widgets(). So it should be noted that these times may not be purely the time take to execute this function if it is preempted. However, after applying this patch and measuring the time taken to execute dapm_power_widgets() again a significant improvement is seen as shown below. Times for function dapm_power_widgets() are (us): Min 4, Ave 16, Max 82, Count 39 Therefore, optimise the dapm_power_widgets() function by only scheduling the dapm_pre/post_sequence_async() work if the DAPM context is not in the desired state. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..2e9231aeabb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1953,7 +1953,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_pre_sequence_async(&card->dapm, 0); /* Run other bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_pre_sequence_async, d, &async_domain); } @@ -1977,7 +1977,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) /* Run all the bias changes in parallel */ list_for_each_entry(d, &card->dapm_list, list) { - if (d != &card->dapm) + if (d != &card->dapm && d->bias_level != d->target_bias_level) async_schedule_domain(dapm_post_sequence_async, d, &async_domain); } -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: q6routing: initialize data correctly
by Srinivas Kandagatla 03 Sep '18

03 Sep '18

Some of the router data fields are left as default zeros which are valid dai ids, so initialize these to invalid value of -1. Without intializing these correctly get_session_from_id() can return incorrect session resulting in not closing the opened copp and messing up with the copp ref count. Fixes: e3a33673e845 ("ASoC: qdsp6: q6routing: Add q6routing driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6routing.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index dc94c5c53788..c6b51571be94 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -960,8 +960,10 @@ static int msm_routing_probe(struct snd_soc_component *c) { int i; - for (i = 0; i < MAX_SESSIONS; i++) + for (i = 0; i < MAX_SESSIONS; i++) { routing_data->sessions[i].port_id = -1; + routing_data->sessions[i].fedai_id = -1; + } return 0; } -- 2.18.0

2 1

[alsa-devel] [PATCH] sound: soc: fix soc-core.c kernel-doc warning
by Randy Dunlap 03 Sep '18

03 Sep '18

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc warning: ../sound/soc/soc-core.c:2918: warning: Excess function parameter 'legacy_dai_naming' description in 'snd_soc_register_dais' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) --- lnx-419-rc2.orig/sound/soc/soc-core.c +++ lnx-419-rc2/sound/soc/soc-core.c @@ -2910,8 +2910,6 @@ static struct snd_soc_dai *soc_add_dai(s * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs - * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the - * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, size_t count)

2 2

[alsa-devel] [PATCH v2 0/2] Add support for MikroElektronika PROTO codec
by Codrin Ciubotariu 03 Sep '18

03 Sep '18

This patchset is based on Florian's initial PROTO driver: https://lkml.org/lkml/2013/5/22/342 My changes made the driver more generic, so that we could connect other I2S CPU DAIs, by using device-tree. Tested with sama5d2 I2S driver. Changes in v2: - removed unnecessary code that added constraints for the fixed SYSLK; - fixed errors reported by kbuild; - adjusted "mikroe" in vendor prefixes; - updated description in Kconfig; Codrin Ciubotariu (2): ASoC: Add driver for PROTO Audio CODEC (with a WM8731) ASoC: mikroe-proto: dt-bindings: add DT bindings for PROTO board .../bindings/sound/mikroe,mikroe-proto.txt | 23 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/atmel/Kconfig | 11 ++ sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mikroe-proto.c | 165 ++++++++++++++++++ 5 files changed, 202 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt create mode 100644 sound/soc/atmel/mikroe-proto.c -- 2.17.1

2 4

[alsa-devel] ASoC: pxa: remove pxa2xx-pcm driver, which caused regression
by petrcvekcz＠gmail.com 02 Sep '18

02 Sep '18

From: Petr Cvek <petrcvekcz(a)gmail.com> The switch of PXA to dmaengine caused a regression for devices which are using SSP/I2S and probably AC97. The problem is in multiple allocation of DMA by function pxa2xx_soc_pcm_new() called by pxa-pcm-audio and (SSP/I2S/AC97). The regression was found on HTC Magician and after playback kernel generates an oops: WARNING: CPU: 0 PID: 892 at drivers/dma/dmaengine.c:788 dma_release_channel+0x44/0xa0 chan reference count 0 != 1 Modules linked in: ... snd_soc_magician snd_soc_uda1380 snd_soc_pxa2xx_i2s spi_pxa2xx_platform snd_soc_pxa2xx snd_soc_pxa_ssp snd_pxa2xx_lib snd_pcm_dmaengine i2c_pxa snd_soc_core snd_pcm snd_timer i2c_core snd ssp soundcore [last unloaded: pwm_bl] CPU: 0 PID: 892 Comm: aplay Tainted: G C 4.18.0-rc6-next-20180726-magician+ #16 Hardware name: HTC Magician [<c0107d90>] (unwind_backtrace) from [<c010594c>] (show_stack+0x10/0x14) [<c010594c>] (show_stack) from [<c0111680>] (__warn+0xd4/0xec) [<c0111680>] (__warn) from [<c01112d0>] (warn_slowpath_fmt+0x44/0x6c) [<c01112d0>] (warn_slowpath_fmt) from [<c034908c>] (dma_release_channel+0x44/0xa0) [<c034908c>] (dma_release_channel) from [<bf330528>] (snd_dmaengine_pcm_close_release_chan+0x44/0xb1c [snd_pcm_dmaengine]) [<bf330528>] (snd_dmaengine_pcm_close_release_chan [snd_pcm_dmaengine]) from [<bf26c29c>] (soc_pcm_components_close+0x8c/0xac [snd_soc_core]) [<bf26c29c>] (soc_pcm_components_close [snd_soc_core]) from [<bf26c3a0>] (soc_pcm_close+0xe4/0xd44 [snd_soc_core]) [<bf26c3a0>] (soc_pcm_close [snd_soc_core]) from [<bf167254>] (snd_pcm_release_substream+0x88/0x108 [snd_pcm]) [<bf167254>] (snd_pcm_release_substream [snd_pcm]) from [<bf167330>] (snd_pcm_release+0x5c/0xd2c [snd_pcm]) [<bf167330>] (snd_pcm_release [snd_pcm]) from [<c01c39e4>] (__fput+0xdc/0x1e8) [<c01c39e4>] (__fput) from [<c0129e7c>] (task_work_run+0xc4/0xd4) [<c0129e7c>] (task_work_run) from [<c01053ac>] (do_work_pending+0xc0/0xc8) [<c01053ac>] (do_work_pending) from [<c0101068>] (slow_work_pending+0xc/0x20) Exception stack(0xc2fdffb0 to 0xc2fdfff8) ffa0: 00000000 0154d1a0 01542a3c 00000000 ffc0: 0154d150 00000000 0009283c 00000006 000003f0 0154d4c8 00000000 0154d4c8 ffe0: b6ee7d70 bef13960 b6e547b8 b6cfe018 60000010 00000004 Removing bindings for snd_soc_magician fixes the problem (tested). Other machines should have the same problem if they use at least one of pxa-ssp-dai or pxa2xx-i2s. Considering pxa2xx-ac97 uses the function pxa2xx_soc_pcm_new() too it should produce the same problem (untested). After removal from I2S/SSP/AC97 machines there seems to be no user of pxa2xx-pcm, so remove the driver completely, change platform device in arch/arm/mach-pxa/devices.c to snd-soc-dummy and an example in the documentation. Signed-off-by: Petr Cvek <petrcvekcz(a)gmail.com> --- Documentation/sound/soc/machine.rst | 2 +- arch/arm/mach-pxa/devices.c | 2 +- sound/soc/pxa/Kconfig | 43 ++++++++++---------------- sound/soc/pxa/Makefile | 2 -- sound/soc/pxa/corgi.c | 1 - sound/soc/pxa/e740_wm9705.c | 2 -- sound/soc/pxa/e750_wm9705.c | 2 -- sound/soc/pxa/e800_wm9712.c | 2 -- sound/soc/pxa/em-x270.c | 2 -- sound/soc/pxa/hx4700.c | 1 - sound/soc/pxa/imote2.c | 1 - sound/soc/pxa/magician.c | 2 -- sound/soc/pxa/mioa701_wm9713.c | 2 -- sound/soc/pxa/palm27x.c | 2 -- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/pxa2xx-pcm.c | 48 ----------------------------- sound/soc/pxa/raumfeld.c | 2 -- sound/soc/pxa/spitz.c | 1 - sound/soc/pxa/tosa.c | 2 -- sound/soc/pxa/z2.c | 1 - sound/soc/pxa/zylonite.c | 3 -- 21 files changed, 18 insertions(+), 106 deletions(-) delete mode 100644 sound/soc/pxa/pxa2xx-pcm.c diff --git a/Documentation/sound/soc/machine.rst b/Documentation/sound/soc/machine.rst index 515c9444deaf..a65f41ade2ca 100644 --- a/Documentation/sound/soc/machine.rst +++ b/Documentation/sound/soc/machine.rst @@ -65,7 +65,7 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. .stream_name = "WM8731", .cpu_dai_name = "pxa-is2-dai", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", + .platform_name = "snd-soc-dummy", .codec_name = "wm8713-codec.0-001a", .init = corgi_wm8731_init, .ops = &corgi_ops, diff --git a/arch/arm/mach-pxa/devices.c b/arch/arm/mach-pxa/devices.c index a24783a03827..82ee7f6c1dcf 100644 --- a/arch/arm/mach-pxa/devices.c +++ b/arch/arm/mach-pxa/devices.c @@ -371,7 +371,7 @@ struct platform_device pxa_device_asoc_ssp4 = { }; struct platform_device pxa_device_asoc_platform = { - .name = "pxa-pcm-audio", + .name = "snd-soc-dummy", .id = -1, }; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 776e148b0aa2..d865c1041c6b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,12 +1,3 @@ -config SND_PXA2XX_SOC - tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA || COMPILE_TEST - select SND_PXA2XX_LIB - help - Say Y or M if you want to add support for codecs attached to - the PXA2xx AC97, I2S or SSP interface. You will also need - to select the audio interfaces to support below. - config SND_MMP_SOC bool "Soc Audio for Marvell MMP chips" depends on ARCH_MMP @@ -43,7 +34,7 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" - depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C + depends on PXA_SHARP_C7xx && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -52,7 +43,7 @@ config SND_PXA2XX_SOC_CORGI config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" - depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C + depends on PXA_SHARP_Cxx00 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -61,7 +52,7 @@ config SND_PXA2XX_SOC_SPITZ config SND_PXA2XX_SOC_Z2 tristate "SoC Audio support for Zipit Z2" - depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C + depends on MACH_ZIPIT2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8750 help @@ -69,7 +60,7 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" - depends on SND_PXA2XX_SOC && MACH_POODLE && I2C + depends on MACH_POODLE && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8731 help @@ -78,7 +69,7 @@ config SND_PXA2XX_SOC_POODLE config SND_PXA2XX_SOC_TOSA tristate "SoC AC97 Audio support for Tosa" - depends on SND_PXA2XX_SOC && MACH_TOSA + depends on MACH_TOSA depends on MFD_TC6393XB select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 @@ -88,7 +79,7 @@ config SND_PXA2XX_SOC_TOSA config SND_PXA2XX_SOC_E740 tristate "SoC AC97 Audio support for e740" - depends on SND_PXA2XX_SOC && MACH_E740 + depends on MACH_E740 select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -97,7 +88,7 @@ config SND_PXA2XX_SOC_E740 config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" - depends on SND_PXA2XX_SOC && MACH_E750 + depends on MACH_E750 select SND_SOC_WM9705 select SND_PXA2XX_SOC_AC97 help @@ -106,7 +97,7 @@ config SND_PXA2XX_SOC_E750 config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" - depends on SND_PXA2XX_SOC && MACH_E800 + depends on MACH_E800 select SND_SOC_WM9712 select SND_PXA2XX_SOC_AC97 help @@ -115,8 +106,7 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ - MACH_CM_X300) + depends on MACH_EM_X270 || MACH_EXEDA || MACH_CM_X300 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -125,8 +115,7 @@ config SND_PXA2XX_SOC_EM_X270 config SND_PXA2XX_SOC_PALM27X bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive" - depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \ - MACH_PALMT5 || MACH_PALMTE2) + depends on MACH_PALMLD || MACH_PALMTX || MACH_PALMT5 || MACH_PALMTE2 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -155,7 +144,7 @@ config SND_SOC_TTC_DKB config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" - depends on SND_PXA2XX_SOC && MACH_ZYLONITE + depends on MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 select SND_PXA_SOC_SSP select SND_SOC_WM9713 @@ -165,7 +154,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" - depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 @@ -175,7 +164,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 && I2C + depends on MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help @@ -184,7 +173,7 @@ config SND_PXA2XX_SOC_HX4700 config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" - depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C + depends on MACH_MAGICIAN && I2C select SND_PXA2XX_SOC_I2S select SND_PXA_SOC_SSP select SND_SOC_UDA1380 @@ -194,7 +183,7 @@ config SND_PXA2XX_SOC_MAGICIAN config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + depends on MACH_MIOA701 select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9713 help @@ -203,7 +192,7 @@ config SND_PXA2XX_SOC_MIOA701 config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C + depends on MACH_INTELMOTE2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5b265662f04f..93c0fdd041ed 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -1,13 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # PXA Platform Support -snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o snd-soc-mmp-objs := mmp-pcm.o snd-soc-mmp-sspa-objs := mmp-sspa.o -obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 054e0d65db9d..d36193e45be9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -265,7 +265,6 @@ static struct snd_soc_dai_link corgi_dai = { .stream_name = "WM8731", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 8ab7032631b7..cfae28571ec7 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -90,7 +90,6 @@ static struct snd_soc_dai_link e740_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, { @@ -98,7 +97,6 @@ static struct snd_soc_dai_link e740_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, }; diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 82bcbbb1841b..21c1f504a141 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -73,7 +73,6 @@ static struct snd_soc_dai_link e750_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9705-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", /* use ops to check startup state */ }, @@ -82,7 +81,6 @@ static struct snd_soc_dai_link e750_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9705-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9705-codec", }, }; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..2475366f0e0b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -74,7 +74,6 @@ static struct snd_soc_dai_link e800_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -82,7 +81,6 @@ static struct snd_soc_dai_link e800_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..76b466b3eaee 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -36,7 +36,6 @@ static struct snd_soc_dai_link em_x270_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -44,7 +43,6 @@ static struct snd_soc_dai_link em_x270_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 6cdef5d4954e..79ffeab616f7 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -144,7 +144,6 @@ static struct snd_soc_dai_link hx4700_dai = { .stream_name = "AK4641", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "ak4641-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "ak4641.0-0012", .init = hx4700_ak4641_init, .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 78475376f971..84dff8d34472 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -51,7 +51,6 @@ static struct snd_soc_dai_link imote2_dai = { .stream_name = "WM8940", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8940-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8940-codec.0-0034", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 935a248e5bf6..a7e7fa08e0a4 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -296,7 +296,6 @@ static struct snd_soc_dai_link magician_dai[] = { .stream_name = "UDA1380 Playback", .cpu_dai_name = "pxa-ssp-dai.0", .codec_dai_name = "uda1380-hifi-playback", - .platform_name = "pxa-pcm-audio", .codec_name = "uda1380-codec.0-0018", .ops = &magician_playback_ops, }, @@ -305,7 +304,6 @@ static struct snd_soc_dai_link magician_dai[] = { .stream_name = "UDA1380 Capture", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "uda1380-hifi-capture", - .platform_name = "pxa-pcm-audio", .codec_name = "uda1380-codec.0-0018", .ops = &magician_capture_ops, } diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 47052fe3f5f4..121dcb1def3c 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -150,7 +150,6 @@ static struct snd_soc_dai_link mioa701_dai[] = { .codec_dai_name = "wm9713-hifi", .codec_name = "wm9713-codec", .init = mioa701_wm9713_init, - .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, }, { @@ -159,7 +158,6 @@ static struct snd_soc_dai_link mioa701_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9713-aux", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .ops = &mioa701_ops, }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..490717c7a5ce 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -94,7 +94,6 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", - .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, { @@ -103,7 +102,6 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .codec_name = "wm9712-codec", - .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index b6693f32fc02..728904266930 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -229,7 +229,6 @@ static struct snd_soc_dai_link poodle_dai = { .stream_name = "WM8731", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c deleted file mode 100644 index 72eaaef1b426..000000000000 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ /dev/null @@ -1,48 +0,0 @@ -/* - * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/dma-mapping.h> -#include <linux/module.h> -#include <linux/dmaengine.h> -#include <linux/of.h> - -#include <sound/core.h> -#include <sound/soc.h> -#include <sound/pxa2xx-lib.h> -#include <sound/dmaengine_pcm.h> - -static const struct snd_soc_component_driver pxa2xx_soc_platform = { - .ops = &pxa2xx_pcm_ops, - .pcm_new = pxa2xx_soc_pcm_new, - .pcm_free = pxa2xx_pcm_free_dma_buffers, -}; - -static int pxa2xx_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_component(&pdev->dev, &pxa2xx_soc_platform, - NULL, 0); -} - -static struct platform_driver pxa_pcm_driver = { - .driver = { - .name = "pxa-pcm-audio", - }, - - .probe = pxa2xx_soc_platform_probe, -}; - -module_platform_driver(pxa_pcm_driver); - -MODULE_AUTHOR("Nicolas Pitre"); -MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:pxa-pcm-audio"); diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 111a907c4eb9..b2ae5aec0b83 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -205,7 +205,6 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .name = "CS4270", \ .stream_name = "CS4270", \ .cpu_dai_name = "pxa-ssp-dai.0", \ - .platform_name = "pxa-pcm-audio", \ .codec_dai_name = "cs4270-hifi", \ .codec_name = "cs4270.0-0048", \ .dai_fmt = SND_SOC_DAIFMT_I2S | \ @@ -220,7 +219,6 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .stream_name = "Playback", \ .cpu_dai_name = "pxa-ssp-dai.1", \ .codec_dai_name = "ak4104-hifi", \ - .platform_name = "pxa-pcm-audio", \ .dai_fmt = SND_SOC_DAIFMT_I2S | \ SND_SOC_DAIFMT_NB_NF | \ SND_SOC_DAIFMT_CBS_CFS, \ diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1671da648e95..57e4ecb9cbe2 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -262,7 +262,6 @@ static struct snd_soc_dai_link spitz_dai = { .stream_name = "WM8750", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..b2ddfeea6b85 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -188,7 +188,6 @@ static struct snd_soc_dai_link tosa_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", .ops = &tosa_ops, }, @@ -197,7 +196,6 @@ static struct snd_soc_dai_link tosa_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", .ops = &tosa_ops, }, diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 5b0eccd2b4dd..5e2307959b7b 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -162,7 +162,6 @@ static struct snd_soc_dai_link z2_dai = { .stream_name = "WM8750", .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8750-hifi", - .platform_name = "pxa-pcm-audio", .codec_name = "wm8750.0-001b", .init = z2_wm8750_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 230eee450f45..753c828ade37 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -132,7 +132,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "AC97", .stream_name = "AC97 HiFi", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9713-hifi", .init = zylonite_wm9713_init, @@ -141,7 +140,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "AC97 Aux", .stream_name = "AC97 Aux", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9713-aux", }, @@ -149,7 +147,6 @@ static struct snd_soc_dai_link zylonite_dai[] = { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", .codec_name = "wm9713-codec", - .platform_name = "pxa-pcm-audio", .cpu_dai_name = "pxa-ssp-dai.2", .codec_dai_name = "wm9713-voice", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 2.18.0

3 10

[alsa-devel] token to prevent spam?
by Lee, Keerock 01 Sep '18

01 Sep '18

Hi all, Could anyone please let me know how to acquire Token to prevent spam when you create an account for the alsa-project.org? I am having trouble in creating an account. Thank you, Keerock

1 0

[alsa-devel] [PATCH] ALSA: control: Make sure that id->index does not overflow in function snd_ctl_replace
by Xiao Yang 01 Sep '18

01 Sep '18

From: Young_X <YangX92(a)hotmail.com> The ALSA control code expects that the range of assigned indices to a control is continuous and does not overflow. Currently there are no checks to enforce this. If a control with a overflowing index range is created that control becomes effectively inaccessible and unremovable since snd_ctl_find_id() will not be able to find it. This patch adds a check that makes sure that controls with a overflowing index range can not be created. (same issue as CVE-2014-4656) Signed-off-by: Young_X <YangX92(a)hotmail.com> --- sound/core/control.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 9aa15bf..6435772 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -441,6 +441,11 @@ int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, goto error; } id = kcontrol->id; + if (id.index > UINT_MAX - kcontrol->count) { + ret = -EINVAL; + goto error; + } + down_write(&card->controls_rwsem); old = snd_ctl_find_id(card, &id); if (!old) { -- 2.7.4

2 1

[alsa-devel] [asoc:for-4.20 333/374] sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'?
by kbuild test robot 01 Sep '18

01 Sep '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.20 head: 5fb6589acc3860304436aa436e7ea33712de6fc2 commit: 6bae5ea9498926440ffc883f3dbceb0adc65e492 [333/374] ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 6bae5ea9498926440ffc883f3dbceb0adc65e492 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sh All errors (new ones prefixed by >>): sound/pci/hda/patch_ca0132.c: In function 'patch_ca0132': >> sound/pci/hda/patch_ca0132.c:7650:20: error: implicit declaration of function 'pci_iomap'; did you mean 'pcim_iomap'? [-Werror=implicit-function-declaration] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^~~~~~~~~ pcim_iomap sound/pci/hda/patch_ca0132.c:7650:18: warning: assignment makes pointer from integer without a cast [-Wint-conversion] spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); ^ cc1: some warnings being treated as errors vim +7650 sound/pci/hda/patch_ca0132.c d5c016b56 Gabriele Martino 2015-05-18 7581 95c6e9cb7 Ian Minett 2011-06-15 7582 static int patch_ca0132(struct hda_codec *codec) 95c6e9cb7 Ian Minett 2011-06-15 7583 { 95c6e9cb7 Ian Minett 2011-06-15 7584 struct ca0132_spec *spec; a73d511c4 Ian Minett 2012-12-20 7585 int err; d5c016b56 Gabriele Martino 2015-05-18 7586 const struct snd_pci_quirk *quirk; 95c6e9cb7 Ian Minett 2011-06-15 7587 4e76a8833 Takashi Iwai 2014-02-25 7588 codec_dbg(codec, "patch_ca0132\n"); 95c6e9cb7 Ian Minett 2011-06-15 7589 95c6e9cb7 Ian Minett 2011-06-15 7590 spec = kzalloc(sizeof(*spec), GFP_KERNEL); 95c6e9cb7 Ian Minett 2011-06-15 7591 if (!spec) 95c6e9cb7 Ian Minett 2011-06-15 7592 return -ENOMEM; 95c6e9cb7 Ian Minett 2011-06-15 7593 codec->spec = spec; 993884f6a Chih-Chung Chang 2013-03-25 7594 spec->codec = codec; 95c6e9cb7 Ian Minett 2011-06-15 7595 225068ab2 Takashi Iwai 2015-05-29 7596 codec->patch_ops = ca0132_patch_ops; 225068ab2 Takashi Iwai 2015-05-29 7597 codec->pcm_format_first = 1; 225068ab2 Takashi Iwai 2015-05-29 7598 codec->no_sticky_stream = 1; 225068ab2 Takashi Iwai 2015-05-29 7599 d5c016b56 Gabriele Martino 2015-05-18 7600 /* Detect codec quirk */ d5c016b56 Gabriele Martino 2015-05-18 7601 quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks); d5c016b56 Gabriele Martino 2015-05-18 7602 if (quirk) d5c016b56 Gabriele Martino 2015-05-18 7603 spec->quirk = quirk->value; d5c016b56 Gabriele Martino 2015-05-18 7604 else d5c016b56 Gabriele Martino 2015-05-18 7605 spec->quirk = QUIRK_NONE; d5c016b56 Gabriele Martino 2015-05-18 7606 e24aa0a4c Takashi Iwai 2014-08-10 7607 spec->dsp_state = DSP_DOWNLOAD_INIT; a7e76271b Ian Minett 2012-12-20 7608 spec->num_mixers = 1; 017310fbe Connor McAdams 2018-05-08 7609 017310fbe Connor McAdams 2018-05-08 7610 /* Set which mixers each quirk uses. */ 017310fbe Connor McAdams 2018-05-08 7611 switch (spec->quirk) { 017310fbe Connor McAdams 2018-05-08 7612 case QUIRK_SBZ: e25e34450 Connor McAdams 2018-08-08 7613 spec->mixers[0] = desktop_mixer; 017310fbe Connor McAdams 2018-05-08 7614 snd_hda_codec_set_name(codec, "Sound Blaster Z"); 017310fbe Connor McAdams 2018-05-08 7615 break; e25e34450 Connor McAdams 2018-08-08 7616 case QUIRK_R3D: e25e34450 Connor McAdams 2018-08-08 7617 spec->mixers[0] = desktop_mixer; e25e34450 Connor McAdams 2018-08-08 7618 snd_hda_codec_set_name(codec, "Recon3D"); e25e34450 Connor McAdams 2018-08-08 7619 break; 017310fbe Connor McAdams 2018-05-08 7620 case QUIRK_R3DI: 017310fbe Connor McAdams 2018-05-08 7621 spec->mixers[0] = r3di_mixer; 017310fbe Connor McAdams 2018-05-08 7622 snd_hda_codec_set_name(codec, "Recon3Di"); 017310fbe Connor McAdams 2018-05-08 7623 break; 017310fbe Connor McAdams 2018-05-08 7624 default: a7e76271b Ian Minett 2012-12-20 7625 spec->mixers[0] = ca0132_mixer; 017310fbe Connor McAdams 2018-05-08 7626 break; 017310fbe Connor McAdams 2018-05-08 7627 } a7e76271b Ian Minett 2012-12-20 7628 08eca6b1f Connor McAdams 2018-08-08 7629 /* Setup whether or not to use alt functions/controls/pci_mmio */ 009b8f979 Connor McAdams 2018-05-08 7630 switch (spec->quirk) { 009b8f979 Connor McAdams 2018-05-08 7631 case QUIRK_SBZ: e42c7c731 Connor McAdams 2018-08-08 7632 case QUIRK_R3D: 08eca6b1f Connor McAdams 2018-08-08 7633 spec->use_alt_controls = true; 08eca6b1f Connor McAdams 2018-08-08 7634 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7635 spec->use_pci_mmio = true; 08eca6b1f Connor McAdams 2018-08-08 7636 break; 009b8f979 Connor McAdams 2018-05-08 7637 case QUIRK_R3DI: 47cdf76e4 Connor McAdams 2018-05-08 7638 spec->use_alt_controls = true; 009b8f979 Connor McAdams 2018-05-08 7639 spec->use_alt_functions = true; 08eca6b1f Connor McAdams 2018-08-08 7640 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7641 break; 009b8f979 Connor McAdams 2018-05-08 7642 default: 47cdf76e4 Connor McAdams 2018-05-08 7643 spec->use_alt_controls = false; 009b8f979 Connor McAdams 2018-05-08 7644 spec->use_alt_functions = false; 08eca6b1f Connor McAdams 2018-08-08 7645 spec->use_pci_mmio = false; 009b8f979 Connor McAdams 2018-05-08 7646 break; 009b8f979 Connor McAdams 2018-05-08 7647 } 009b8f979 Connor McAdams 2018-05-08 7648 08eca6b1f Connor McAdams 2018-08-08 7649 if (spec->use_pci_mmio) { 08eca6b1f Connor McAdams 2018-08-08 @7650 spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); 08eca6b1f Connor McAdams 2018-08-08 7651 if (spec->mem_base == NULL) { 08eca6b1f Connor McAdams 2018-08-08 7652 codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE."); 08eca6b1f Connor McAdams 2018-08-08 7653 spec->quirk = QUIRK_NONE; 08eca6b1f Connor McAdams 2018-08-08 7654 } 08eca6b1f Connor McAdams 2018-08-08 7655 } 08eca6b1f Connor McAdams 2018-08-08 7656 5aaca44d8 Ian Minett 2012-12-20 7657 spec->base_init_verbs = ca0132_base_init_verbs; 5aaca44d8 Ian Minett 2012-12-20 7658 spec->base_exit_verbs = ca0132_base_exit_verbs; 5aaca44d8 Ian Minett 2012-12-20 7659 993884f6a Chih-Chung Chang 2013-03-25 7660 INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); 993884f6a Chih-Chung Chang 2013-03-25 7661 95c6e9cb7 Ian Minett 2011-06-15 7662 ca0132_init_chip(codec); 95c6e9cb7 Ian Minett 2011-06-15 7663 95c6e9cb7 Ian Minett 2011-06-15 7664 ca0132_config(codec); 95c6e9cb7 Ian Minett 2011-06-15 7665 d5c016b56 Gabriele Martino 2015-05-18 7666 err = ca0132_prepare_verbs(codec); d5c016b56 Gabriele Martino 2015-05-18 7667 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7668 goto error; d5c016b56 Gabriele Martino 2015-05-18 7669 a73d511c4 Ian Minett 2012-12-20 7670 err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); a73d511c4 Ian Minett 2012-12-20 7671 if (err < 0) cc91ceaf3 Takashi Iwai 2017-09-04 7672 goto error; a73d511c4 Ian Minett 2012-12-20 7673 95c6e9cb7 Ian Minett 2011-06-15 7674 return 0; cc91ceaf3 Takashi Iwai 2017-09-04 7675 cc91ceaf3 Takashi Iwai 2017-09-04 7676 error: cc91ceaf3 Takashi Iwai 2017-09-04 7677 ca0132_free(codec); cc91ceaf3 Takashi Iwai 2017-09-04 7678 return err; 95c6e9cb7 Ian Minett 2011-06-15 7679 } 95c6e9cb7 Ian Minett 2011-06-15 7680 :::::: The code at line 7650 was first introduced by commit :::::: 08eca6b1f1468a4021bac7b3929fd3eb491e2629 ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio :::::: TO: Connor McAdams <conmanx360(a)gmail.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] Applied "ALSA: hda: move hda_codec.h to include/sound" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ALSA: hda: move hda_codec.h to include/sound has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From be57bfffb7b5ba72e1293643053f8861fcaa5163 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 22 Aug 2018 15:24:57 -0500 Subject: [PATCH] ALSA: hda: move hda_codec.h to include/sound As suggested by Takashi, move this header file to make it easier to include from e.g. the Intel Skylake driver in follow-up patches Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- {sound/pci/hda => include/sound}/hda_codec.h | 0 sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/hda_bind.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- 25 files changed, 24 insertions(+), 24 deletions(-) rename {sound/pci/hda => include/sound}/hda_codec.h (100%) diff --git a/sound/pci/hda/hda_codec.h b/include/sound/hda_codec.h similarity index 100% rename from sound/pci/hda/hda_codec.h rename to include/sound/hda_codec.h diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b9a6b66aeb0e..df0d636145f8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -13,7 +13,7 @@ #include <linux/export.h> #include <linux/sort.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index d1a6a9c1329a..f1457c6b3969 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -9,7 +9,7 @@ #ifndef __SOUND_HDA_BEEP_H #define __SOUND_HDA_BEEP_H -#include "hda_codec.h" +#include <sound/hda_codec.h> #define HDA_BEEP_MODE_OFF 0 #define HDA_BEEP_MODE_ON 1 diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d361bb77ca00..2222b47d4ec4 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -11,7 +11,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0a5085537034..ccbb0d12b8cc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -27,7 +27,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/asoundef.h> #include <sound/tlv.h> #include <sound/initval.h> diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index a68e75b00ea3..55760e5231e6 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -20,7 +20,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/hda_register.h> #define AZX_MAX_CODECS HDA_MAX_CODECS diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index ba7fe9b6655c..806b12ed44a2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <asm/unaligned.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" enum eld_versions { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 579984ecdec3..276150f29cda 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -32,7 +32,7 @@ #include <sound/core.h> #include <sound/jack.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index cc009a4a3d1d..268bba6ec985 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -23,7 +23,7 @@ #include <linux/compat.h> #include <linux/nospec.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..c8fde95e2393 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -63,7 +63,7 @@ #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" #include "hda_intel.h" diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index a33234e04d4f..c499727920e6 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -15,7 +15,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c6b778b2580c..a65740419650 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <sound/core.h> #include <linux/module.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" static int dump_coef = -1; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 6ec79c58d48d..c154b19a0c45 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -14,7 +14,7 @@ #include <linux/string.h> #include <linux/export.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0621920f7617..4bc5232eac1c 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -35,7 +35,7 @@ #include <sound/core.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" /* Defines for Nvidia Tegra HDA support */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fd476fb40e1b..ebfd0be885b3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -24,7 +24,7 @@ #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index c2d9ee9cfdc0..21d0f0610913 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -22,7 +22,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0166a3d7cd55..a585d0ec6d77 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -32,7 +32,7 @@ #include <linux/io.h> #include <linux/pci.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a7f91be45194..64fa5a82bb9f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -23,7 +23,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 1b2195dd2b26..52642ba3e2c0 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cfd4e4f97f8f..5592557fe50e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cb587dce67a9..67099cbb6be2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -41,7 +41,7 @@ #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d117f00d04d..6f3c8e888c2a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -32,7 +32,7 @@ #include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index f63acb1b965c..c49d25bcd7f2 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -27,7 +27,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* si3054 verbs */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 046705b4691a..d16a25a395c9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,7 +32,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 6b9617aee0e6..9f6f13e25145 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -52,7 +52,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/asoundef.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add support for FIFO usage caused delay reporting
by Peter Ujfalusi 31 Aug '18

31 Aug '18

McASP have write and read FIFO, each 64 words deep. >From the WFIFOS/RFIFOS registers we can read the amount of data currently in the FIFO which can be directly reported as delay. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 37 +++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f7f546bdd312..1f8f4f3929f2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1062,6 +1062,42 @@ static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, return error_ppm; } +static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->txnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET); +} + +static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp) +{ + if (!mcasp->rxnumevt) + return 0; + + return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET); +} + +static snd_pcm_sframes_t davinci_mcasp_delay( + struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + u32 fifo_use; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_use = davinci_mcasp_tx_delay(mcasp); + else + fifo_use = davinci_mcasp_rx_delay(mcasp); + + /* + * Divide the used locations with the channel count to get the + * FIFO usage in samples (don't care about partial samples in the + * buffer). + */ + return fifo_use / substream->runtime->channels; +} + static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -1386,6 +1422,7 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .startup = davinci_mcasp_startup, .shutdown = davinci_mcasp_shutdown, .trigger = davinci_mcasp_trigger, + .delay = davinci_mcasp_delay, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, .set_clkdiv = davinci_mcasp_set_clkdiv, -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: codecs: tas5720: add TAS5722 specific volume control
by Andrew F. Davis 31 Aug '18

31 Aug '18

From: Andreas Dannenberg <dannenberg(a)ti.com> The TAS5722 supports modifying volume in 0.25dB steps (as opposed to 0.5dB steps on the TAS5720). Introduce a custom mixer control that allows taking advantage of this finer output volume granularity. Also add custom getters/setters for access as the TAS5722 digital volume controls are split over two registers. Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Revise commit message - codec -> component sound/soc/codecs/tas5720.c | 88 ++++++++++++++++++++++++++++++++++---- 1 file changed, 80 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index ae3d032ac35a..3c469112477a 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -485,15 +485,56 @@ static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, ); /* - * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that - * setting the gain below -100 dB (register value <0x7) is effectively a MUTE - * as per device datasheet. + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB or 0.25 dB steps + * depending on the device. Note that setting the gain below -100 dB + * (register value <0x7) is effectively a MUTE as per device datasheet. + * + * Note that for the TAS5722 the digital volume controls are actually split + * over two registers, so we need custom getters/setters for access. */ -static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5720_dac_tlv, -10350, 50, 0); +static DECLARE_TLV_DB_SCALE(tas5722_dac_tlv, -10350, 25, 0); + +static int tas5722_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int val; + + snd_soc_component_read(component, TAS5720_VOLUME_CTRL_REG, &val); + ucontrol->value.integer.value[0] = val << 1; + + snd_soc_component_read(component, TAS5722_DIGITAL_CTRL2_REG, &val); + ucontrol->value.integer.value[0] |= val & TAS5722_VOL_CONTROL_LSB; + + return 0; +} + +static int tas5722_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + unsigned int sel = ucontrol->value.integer.value[0]; + + snd_soc_component_write(component, TAS5720_VOLUME_CTRL_REG, sel >> 1); + snd_soc_component_update_bits(component, TAS5722_DIGITAL_CTRL2_REG, + TAS5722_VOL_CONTROL_LSB, sel); + + return 0; +} static const struct snd_kcontrol_new tas5720_snd_controls[] = { SOC_SINGLE_TLV("Speaker Driver Playback Volume", - TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, tas5720_dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_kcontrol_new tas5722_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Speaker Driver Playback Volume", + 0, 0, 511, 0, + tas5722_volume_get, tas5722_volume_set, + tas5722_dac_tlv), SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), }; @@ -527,6 +568,23 @@ static const struct snd_soc_component_driver soc_component_dev_tas5720 = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver soc_component_dev_tas5722 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + .controls = tas5722_snd_controls, + .num_controls = ARRAY_SIZE(tas5722_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + /* PCM rates supported by the TAS5720 driver */ #define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) @@ -613,9 +671,23 @@ static int tas5720_probe(struct i2c_client *client, dev_set_drvdata(dev, data); - ret = devm_snd_soc_register_component(&client->dev, - &soc_component_dev_tas5720, - tas5720_dai, ARRAY_SIZE(tas5720_dai)); + switch (id->driver_data) { + case TAS5720: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5720, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + case TAS5722: + ret = devm_snd_soc_register_component(&client->dev, + &soc_component_dev_tas5722, + tas5720_dai, + ARRAY_SIZE(tas5720_dai)); + break; + default: + dev_err(dev, "unexpected private driver data\n"); + return -EINVAL; + } if (ret < 0) { dev_err(dev, "failed to register component: %d\n", ret); return ret; -- 2.18.0

2 3

[alsa-devel] Applied "ASoC: cs4265: SOC_SINGLE register value error fix" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: cs4265: SOC_SINGLE register value error fix has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6f18bcdaa24bae39c746b57b95af19ff3c41b17f Mon Sep 17 00:00:00 2001 From: Matt Flax <flatmax(a)flatmax.org> Date: Thu, 30 Aug 2018 09:38:00 +1000 Subject: [PATCH] ASoC: cs4265: SOC_SINGLE register value error fix The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de669f..15b4ae04870f 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: cs4265: Add native 32bit I2S transport" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: cs4265: Add native 32bit I2S transport has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From be47e75eb1419ffc1d9c26230963fd5fa3055097 Mon Sep 17 00:00:00 2001 From: Matt Flax <flatmax(a)flatmax.org> Date: Thu, 30 Aug 2018 09:38:01 +1000 Subject: [PATCH] ASoC: cs4265: Add native 32bit I2S transport The cs4265 uses 32 bit transport on the I2S bus. This patch enables native 32 bit mode for machine drivers which use this sound card driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs4265.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 15b4ae04870f..17d7e6f0dcdb 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -495,7 +495,8 @@ static int cs4265_set_bias_level(struct snd_soc_component *component, SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) static const struct snd_soc_dai_ops cs4265_ops = { .hw_params = cs4265_pcm_hw_params, -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: cs4265: SOC_SINGLE register value error fix
by Matt Flax 31 Aug '18

31 Aug '18

The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- Acked-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Identical to the original v1 patch, already acked and reviewed. --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..15b4ae04 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.17.1

3 7

[alsa-devel] Applied "ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e664de680b10c13d65f982bcb9cfe56096e1de55 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:08:09 +0000 Subject: [PATCH] ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple_card_utils for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/simple_card_utils.h | 27 +++++++++++------ sound/soc/generic/simple-card-utils.c | 42 +++++++++++++++++++++++++-- 2 files changed, 58 insertions(+), 11 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 8bc5e2d8b13c..3b5bd6e76f88 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -51,29 +51,35 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, #define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai) \ asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \ - dai_link->cpu_dai_name) + dai_link->cpu_dai_name, NULL) #define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai) \ asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\ - dai_link->codec_dai_name) + dai_link->codec_dai_name, dai_link->codecs) int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, - const char *name); + const char *dai_name, + struct snd_soc_dai_link_component *dlc); int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ - asoc_simple_card_parse_dai(node, &dai_link->cpu_of_node, \ + asoc_simple_card_parse_dai(node, NULL, \ + &dai_link->cpu_of_node, \ &dai_link->cpu_dai_name, list_name, cells_name, is_single_link) #define asoc_simple_card_parse_codec(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, &dai_link->codec_of_node, \ - &dai_link->codec_dai_name, list_name, cells_name, NULL) + asoc_simple_card_parse_dai(node, dai_link->codecs, \ + &dai_link->codec_of_node, \ + &dai_link->codec_dai_name, \ + list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, &dai_link->platform_of_node, \ + asoc_simple_card_parse_dai(node, NULL, \ + &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, struct device_node **endpoint_np, const char **dai_name, const char *list_name, @@ -81,12 +87,15 @@ int asoc_simple_card_parse_dai(struct device_node *node, int *is_single_links); #define asoc_simple_card_parse_graph_cpu(ep, dai_link) \ - asoc_simple_card_parse_graph_dai(ep, &dai_link->cpu_of_node, \ + asoc_simple_card_parse_graph_dai(ep, NULL, \ + &dai_link->cpu_of_node, \ &dai_link->cpu_dai_name) #define asoc_simple_card_parse_graph_codec(ep, dai_link) \ - asoc_simple_card_parse_graph_dai(ep, &dai_link->codec_of_node, \ + asoc_simple_card_parse_graph_dai(ep, dai_link->codecs, \ + &dai_link->codec_of_node, \ &dai_link->codec_dai_name) int asoc_simple_card_parse_graph_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, struct device_node **endpoint_np, const char **dai_name); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d3f3f0fec74c..73c0a904f32e 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -173,11 +173,23 @@ int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai, - const char *name) + const char *dai_name, + struct snd_soc_dai_link_component *dlc) { struct clk *clk; u32 val; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_of_node = dlc->of_node; + dai_name = dlc->dai_name; + } + /* * Parse dai->sysclk come from "clocks = <&xxx>" * (if system has common clock) @@ -200,7 +212,7 @@ int asoc_simple_card_parse_clk(struct device *dev, if (of_property_read_bool(node, "system-clock-direction-out")) simple_dai->clk_direction = SND_SOC_CLOCK_OUT; - dev_dbg(dev, "%s : sysclk = %d, direction %d\n", name, + dev_dbg(dev, "%s : sysclk = %d, direction %d\n", dai_name, simple_dai->sysclk, simple_dai->clk_direction); return 0; @@ -208,6 +220,7 @@ int asoc_simple_card_parse_clk(struct device *dev, EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); int asoc_simple_card_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, const char **dai_name, const char *list_name, @@ -220,6 +233,17 @@ int asoc_simple_card_parse_dai(struct device_node *node, if (!node) return 0; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + /* * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() @@ -278,6 +302,7 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) } int asoc_simple_card_parse_graph_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, const char **dai_name) { @@ -285,6 +310,17 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, struct of_phandle_args args; int ret; + /* + * Use snd_soc_dai_link_component instead of legacy style. + * It is only for codec, but cpu will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + if (dlc) { + dai_name = &dlc->dai_name; + dai_of_node = &dlc->of_node; + } + if (!ep) return 0; if (!dai_name) @@ -374,6 +410,8 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) num_links++, dai_link++) { of_node_put(dai_link->cpu_of_node); of_node_put(dai_link->codec_of_node); + if (dai_link->codecs) + of_node_put(dai_link->codecs->of_node); } return 0; } -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH v2 00/15] ASoC: use snd_soc_dai_link_component style for simple card
by Kuninori Morimoto 31 Aug '18

31 Aug '18

Hi Mark These are v2 patch of snd_soc_dai_link_component style conversion for simple/audo card driver. The difference between v1 and v2 was "platform series" was missing code for DT, and tidyuped unreadable code. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [we want] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Kuninori Morimoto (15): ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec ASoC: simple-card: support snd_soc_dai_link_component style for codec ASoC: simple-scu-card: use simple_dai_props ASoC: simple-scu-card: support snd_soc_dai_link_component style for codec ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec ASoC: audio-graph-scu-card: use simple_dai_props ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for codec ASoC: simple-card-util: remove dai_link compatible code for codec ASoC: soc-core: use snd_soc_dai_link_component for platform ASoC: simple-card-util: support snd_soc_dai_link_component style for platform ASoC: simple-card: support snd_soc_dai_link_component style for platform ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform ASoC: simple-card-util: remove dai_link compatible code for platform include/sound/simple_card_utils.h | 27 ++++++++++------ include/sound/soc.h | 2 ++ sound/soc/generic/audio-graph-card.c | 18 +++++++++-- sound/soc/generic/audio-graph-scu-card.c | 55 +++++++++++++++++++++----------- sound/soc/generic/simple-card-utils.c | 47 ++++++++++++++++++++++++--- sound/soc/generic/simple-card.c | 30 ++++++++++++++--- sound/soc/generic/simple-scu-card.c | 54 ++++++++++++++++++++----------- sound/soc/soc-core.c | 48 ++++++++++++++++++++++++---- 8 files changed, 217 insertions(+), 64 deletions(-) -- 2.7.4

2 20

[alsa-devel] Applied "ASoC: simple-scu-card: use simple_dai_props" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: simple-scu-card: use simple_dai_props has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5ece10ab99205d11419e3d2e6c7ac0e382d952b5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:08:38 +0000 Subject: [PATCH] ASoC: simple-scu-card: use simple_dai_props simple-card and simple-scu-card are very similar driver, but using different feature. Thus we are keeping synchronization on these 2 drivers style, because it is easy to confirm / check. Current big difference between these 2 drivers are "dai_props" on simple_card_data (= priv). It will be difficult to keep synchronize if we will add new feature on simple-scu-card. Thus, this patch synchronize it. [simple] struct simple_card_data { ... struct simple_dai_props { ... } *dai_props; ... }; [simple scu] struct simple_card_data { ... struct asoc_simple_dai *dai_props; ... }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 16a83bc51e0e..09be02e7416f 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -22,7 +22,9 @@ struct simple_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; - struct asoc_simple_dai *dai_props; + struct simple_dai_props { + struct asoc_simple_dai dai; + } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; }; @@ -40,20 +42,20 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - return asoc_simple_card_clk_enable(dai_props); + return asoc_simple_card_clk_enable(&dai_props->dai); } static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props); + asoc_simple_card_clk_disable(&dai_props->dai); } static const struct snd_soc_ops asoc_simple_card_ops = { @@ -66,7 +68,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct simple_dai_props *dai_props; int num = rtd->num; dai_link = simple_priv_to_link(priv, num); @@ -75,7 +77,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) rtd->cpu_dai : rtd->codec_dai; - return asoc_simple_card_init_dai(dai, dai_props); + return asoc_simple_card_init_dai(dai, &dai_props->dai); } static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -95,7 +97,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); - struct asoc_simple_dai *dai_props = simple_priv_to_props(priv, idx); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct snd_soc_card *card = simple_priv_to_card(priv); int ret; @@ -116,7 +118,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -141,7 +143,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -157,7 +159,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, PREFIX "prefix"); } - ret = asoc_simple_card_of_parse_tdm(np, dai_props); + ret = asoc_simple_card_of_parse_tdm(np, &dai_props->dai); if (ret) return ret; @@ -230,7 +232,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) { struct simple_card_data *priv; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct simple_dai_props *dai_props; struct snd_soc_card *card; struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8e6746db2e66c30e59a49ac15eb0c54d51acfb4b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:09:05 +0000 Subject: [PATCH] ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-card for codec. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2094d2c8919f..5b2ecf8c2652 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -25,6 +25,7 @@ struct graph_card_data { struct graph_dai_props { struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; + struct snd_soc_dai_link_component codecs; /* single codec */ unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -213,7 +214,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *cpu_port, ret = asoc_simple_card_set_dailink_name(dev, dai_link, "%s-%s", dai_link->cpu_dai_name, - dai_link->codec_dai_name); + dai_link->codecs->dai_name); if (ret < 0) goto dai_link_of_err; @@ -299,7 +300,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) struct graph_dai_props *dai_props; struct device *dev = &pdev->dev; struct snd_soc_card *card; - int num, ret; + int num, ret, i; /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -315,6 +316,17 @@ static int asoc_graph_card_probe(struct platform_device *pdev) if (!dai_props || !dai_link) return -ENOMEM; + /* + * Use snd_soc_dai_link_component instead of legacy style + * It is codec only. but cpu/platform will be supported in the future. + * see + * soc-core.c :: snd_soc_init_multicodec() + */ + for (i = 0; i < num; i++) { + dai_link[i].codecs = &dai_props[i].codecs; + dai_link[i].num_codecs = 1; + } + priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); if (IS_ERR(priv->pa_gpio)) { ret = PTR_ERR(priv->pa_gpio); -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: use simple_dai_props" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: audio-graph-scu-card: use simple_dai_props has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1340739d4de4d9d99f1134180f95b42cc4eda438 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:09:20 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: use simple_dai_props audi-graph-card and audio-graph-scu-card are very similar driver, but using different feature. Thus we are keeping synchronization on these 2 drivers style, because it is easy to confirm / check. Current big difference between these 2 drivers are "dai_props" on graph_card_data (= priv). It will be difficult to keep synchronize if we will add new feature on audio-graph-scu-card. Thus, this patch synchronize it. [audio-graph] struct graph_card_data { ... struct graph_dai_props { ... } *dai_props; ... }; [audio-graph-scu] struct graph_card_data { ... struct asoc_simple_dai *dai_props; ... }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-scu-card.c | 26 +++++++++++++----------- 1 file changed, 14 insertions(+), 12 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index 92882e392d6c..043938fece42 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -25,7 +25,9 @@ struct graph_card_data { struct snd_soc_card snd_card; struct snd_soc_codec_conf codec_conf; - struct asoc_simple_dai *dai_props; + struct graph_dai_props { + struct asoc_simple_dai dai; + } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; }; @@ -39,18 +41,18 @@ static int asoc_graph_card_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - return asoc_simple_card_clk_enable(dai_props); + return asoc_simple_card_clk_enable(&dai_props->dai); } static void asoc_graph_card_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, rtd->num); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props); + asoc_simple_card_clk_disable(&dai_props->dai); } static const struct snd_soc_ops asoc_graph_card_ops = { @@ -63,7 +65,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) struct graph_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct graph_dai_props *dai_props; int num = rtd->num; dai_link = graph_priv_to_link(priv, num); @@ -72,7 +74,7 @@ static int asoc_graph_card_dai_init(struct snd_soc_pcm_runtime *rtd) rtd->cpu_dai : rtd->codec_dai; - return asoc_simple_card_init_dai(dai, dai_props); + return asoc_simple_card_init_dai(dai, &dai_props->dai); } static int asoc_graph_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -92,7 +94,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, { struct device *dev = graph_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, idx); - struct asoc_simple_dai *dai_props = graph_priv_to_props(priv, idx); + struct graph_dai_props *dai_props = graph_priv_to_props(priv, idx); struct snd_soc_card *card = graph_priv_to_card(priv); int ret; @@ -110,7 +112,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -137,7 +139,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, &dai_props->dai); if (ret < 0) return ret; @@ -153,7 +155,7 @@ static int asoc_graph_card_dai_link_of(struct device_node *ep, "prefix"); } - ret = asoc_simple_card_of_parse_tdm(ep, dai_props); + ret = asoc_simple_card_of_parse_tdm(ep, &dai_props->dai); if (ret) return ret; @@ -331,7 +333,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) { struct graph_card_data *priv; struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dai_props; + struct graph_dai_props *dai_props; struct device *dev = &pdev->dev; struct snd_soc_card *card; int num, ret; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: simple-card-util: remove dai_link compatible code for codec" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: simple-card-util: remove dai_link compatible code for codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2967e5ea19ec041104fd43da4cb07e8e76a39e55 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:09:47 +0000 Subject: [PATCH] ASoC: simple-card-util: remove dai_link compatible code for codec Now no simple/audio cards are using legacy dai_link style for codec. Let's remove compatible code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 73c0a904f32e..e7057be957e1 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -409,9 +409,7 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) num_links < card->num_links; num_links++, dai_link++) { of_node_put(dai_link->cpu_of_node); - of_node_put(dai_link->codec_of_node); - if (dai_link->codecs) - of_node_put(dai_link->codecs->of_node); + of_node_put(dai_link->codecs->of_node); } return 0; } -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: soc-core: use snd_soc_dai_link_component for platform" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: soc-core: use snd_soc_dai_link_component for platform has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From daecf46ee0e5f0fb2349e20af53c4653e2afc440 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:10:08 +0000 Subject: [PATCH] ASoC: soc-core: use snd_soc_dai_link_component for platform Current struct snd_soc_dai_link is supporting multicodec, and it is supporting legacy style of codec_name codec_of_node code_dai_name This is handled as single entry of multicodec. We don't have multicpu support yet, but in the future we will. In such case, we can use snd_soc_dai_link_component for both cpu/codec. Then the code will be more simple and readble. As next step, we want to use it for platform, too. This patch adds snd_soc_dai_link_component style for platform. We might have multiplatform support in the future, but we don't know yet. To avoid un-known issue / complex code, this patch supports just single-platform as 1st step. If we could use snd_soc_dai_link_component for all CPU/Codec/Platform, we will switch to new style, and remove legacy code. This is prepare for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 48 +++++++++++++++++++++++++++++++++++++------- 2 files changed, 43 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 41cec42fb456..96c19aabf21b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -915,6 +915,8 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; + struct snd_soc_dai_link_component *platform; + int id; /* optional ID for machine driver link identification */ const struct snd_soc_pcm_stream *params; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 473eefe8658e..2a73630d0680 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -892,8 +892,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->codec_dai = codec_dais[0]; /* if there's no platform we match on the empty platform */ - platform_name = dai_link->platform_name; - if (!platform_name && !dai_link->platform_of_node) + platform_name = dai_link->platform->name; + if (!platform_name && !dai_link->platform->of_node) platform_name = "snd-soc-dummy"; /* find one from the set of registered platforms */ @@ -902,8 +902,8 @@ static int soc_bind_dai_link(struct snd_soc_card *card, if (!platform_of_node && component->dev->parent->of_node) platform_of_node = component->dev->parent->of_node; - if (dai_link->platform_of_node) { - if (platform_of_node != dai_link->platform_of_node) + if (dai_link->platform->of_node) { + if (platform_of_node != dai_link->platform->of_node) continue; } else { if (strcmp(component->name, platform_name)) @@ -1015,6 +1015,31 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static int snd_soc_init_platform(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* + * FIXME + * + * this function should be removed in the future + */ + /* convert Legacy platform link */ + if (dai_link->platform) + return 0; + + dai_link->platform = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->platform) + return -ENOMEM; + + dai_link->platform->name = dai_link->platform_name; + dai_link->platform->of_node = dai_link->platform_of_node; + dai_link->platform->dai_name = NULL; + + return 0; +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -1047,6 +1072,12 @@ static int soc_init_dai_link(struct snd_soc_card *card, { int i, ret; + ret = snd_soc_init_platform(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multiplatform\n"); + return ret; + } + ret = snd_soc_init_multicodec(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init multicodec\n"); @@ -1076,13 +1107,12 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform_name && link->platform_of_node) { + if (link->platform->name && link->platform->of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); return -EINVAL; } - /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1917,7 +1947,11 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) card->dai_link[i].name); /* override platform component */ - dai_link->platform_name = component->name; + if (snd_soc_init_platform(card, dai_link) < 0) { + dev_err(card->dev, "init platform error"); + continue; + } + dai_link->platform->name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 24f3bead9b72cbd91956be28ab6a74d51bdd4c6b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:10:46 +0000 Subject: [PATCH] ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for simple-scu-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 91efc8653746..85b46f0eae0f 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -25,6 +25,7 @@ struct simple_card_data { struct simple_dai_props { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; + struct snd_soc_dai_link_component platform; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -262,6 +263,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->dai_props = dai_props; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 46c73187f2986e40b427485bb8f4401aa9143ed0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:10:58 +0000 Subject: [PATCH] ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 5b2ecf8c2652..fb6635f8d5d7 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -26,6 +26,7 @@ struct graph_card_data { struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ + struct snd_soc_dai_link_component platform; unsigned int mclk_fs; } *dai_props; unsigned int mclk_fs; @@ -325,6 +326,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 77b9b84132f0b9ca0802a25277eb7be49713661f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:11:12 +0000 Subject: [PATCH] ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform Current ASoC is supporting snd_soc_dai_link_component for binding, it is more useful than current legacy style. Currently only codec is supporting it as multicodec (= codecs). CPU will support multi style in the future. We want to have it on Platform too in the future. If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component style, we can remove legacy complex style. This patch supports snd_soc_dai_link_component style for audio-graph-scu-card for platform. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [in the future] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-scu-card.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index eeb3c1975fe3..b83bb31021a9 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -28,6 +28,7 @@ struct graph_card_data { struct graph_dai_props { struct asoc_simple_dai dai; struct snd_soc_dai_link_component codecs; + struct snd_soc_dai_link_component platform; } *dai_props; struct snd_soc_dai_link *dai_link; struct asoc_simple_card_data adata; @@ -365,6 +366,7 @@ static int asoc_graph_card_probe(struct platform_device *pdev) for (i = 0; i < num; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; + dai_link[i].platform = &dai_props[i].platform; } priv->dai_props = dai_props; -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: simple-card-util: remove dai_link compatible code for platform" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: simple-card-util: remove dai_link compatible code for platform has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c2f0898b86486a459fa8c91d194f3b699498c0c1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Fri, 31 Aug 2018 03:11:25 +0000 Subject: [PATCH] ASoC: simple-card-util: remove dai_link compatible code for platform Now no simple/audio cards are using legacy dai_link style for platform. Let's remove compatible code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card-utils.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 644cd62ba027..b400dbf1f834 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -376,13 +376,9 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (dai_link->platform) { - if (!dai_link->platform->of_node) - dai_link->platform->of_node = dai_link->cpu_of_node; - } else { - if (!dai_link->platform_of_node) - dai_link->platform_of_node = dai_link->cpu_of_node; - } + if (!dai_link->platform->of_node) + dai_link->platform->of_node = dai_link->cpu_of_node; + return 0; } -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: tas6424: Print full register name in error message" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: tas6424: Print full register name in error message has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 919869214b8e0b24926a278e121879f60df485bb Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Fri, 31 Aug 2018 10:14:06 -0500 Subject: [PATCH] ASoC: tas6424: Print full register name in error message The current short version of the register name may be ambiguous when another fault register detection is added. Use the full name. While here fix comment about clearing faults, the CLEAR_FAULT register actually only clears sticky bits, which are only warnings, fault bits can only cleared by resolving the fault. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas6424.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 14999b999fd3..3315ce8a15e6 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -408,7 +408,7 @@ static void tas6424_fault_check_work(struct work_struct *work) ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg); if (ret < 0) { - dev_err(dev, "failed to read FAULT1 register: %d\n", ret); + dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret); goto out; } @@ -451,7 +451,7 @@ static void tas6424_fault_check_work(struct work_struct *work) check_global_fault2_reg: ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT2, &reg); if (ret < 0) { - dev_err(dev, "failed to read FAULT2 register: %d\n", ret); + dev_err(dev, "failed to read GLOB_FAULT2 register: %d\n", ret); goto out; } @@ -524,7 +524,7 @@ static void tas6424_fault_check_work(struct work_struct *work) /* Store current warn value so we can detect any changes next time */ tas6424->last_warn = reg; - /* Clear any faults by toggling the CLEAR_FAULT control bit */ + /* Clear any warnings by toggling the CLEAR_FAULT control bit */ ret = regmap_write_bits(tas6424->regmap, TAS6424_MISC_CTRL3, TAS6424_CLEAR_FAULT, TAS6424_CLEAR_FAULT); if (ret < 0) -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: tas6424: Save last fault register even when clear" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: tas6424: Save last fault register even when clear has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d40e3e9e44db4b3c8777f3b515ba6097ba26e3b2 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Fri, 31 Aug 2018 10:14:05 -0500 Subject: [PATCH] ASoC: tas6424: Save last fault register even when clear When there is no fault bit set in a fault register we skip the fault reporting section for that register. This also skips over saving that registers value. We save the value so we will not double report an error, but if an error clears then returns we will also not report it as we did not save the all cleared register value. Fix this by saving the fault register value in the all clear path. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/tas6424.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 14999b999fd3..0d6145549a98 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -424,8 +424,10 @@ static void tas6424_fault_check_work(struct work_struct *work) TAS6424_FAULT_PVDD_UV | TAS6424_FAULT_VBAT_UV; - if (reg) + if (!reg) { + tas6424->last_fault1 = reg; goto check_global_fault2_reg; + } /* * Only flag errors once for a given occurrence. This is needed as @@ -461,8 +463,10 @@ static void tas6424_fault_check_work(struct work_struct *work) TAS6424_FAULT_OTSD_CH3 | TAS6424_FAULT_OTSD_CH4; - if (!reg) + if (!reg) { + tas6424->last_fault2 = reg; goto check_warn_reg; + } if ((reg & TAS6424_FAULT_OTSD) && !(tas6424->last_fault2 & TAS6424_FAULT_OTSD)) dev_crit(dev, "experienced a global overtemp shutdown\n"); @@ -497,8 +501,10 @@ static void tas6424_fault_check_work(struct work_struct *work) TAS6424_WARN_VDD_OTW_CH3 | TAS6424_WARN_VDD_OTW_CH4; - if (!reg) + if (!reg) { + tas6424->last_warn = reg; goto out; + } if ((reg & TAS6424_WARN_VDD_UV) && !(tas6424->last_warn & TAS6424_WARN_VDD_UV)) dev_warn(dev, "experienced a VDD under voltage condition\n"); -- 2.19.0.rc1

1 0

[alsa-devel] Applied "ASoC: tas6424: Add channel fault reporting" to the asoc tree
by Mark Brown 31 Aug '18

31 Aug '18

The patch ASoC: tas6424: Add channel fault reporting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5fb6589acc3860304436aa436e7ea33712de6fc2 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Fri, 31 Aug 2018 10:14:07 -0500 Subject: [PATCH] ASoC: tas6424: Add channel fault reporting The TAS6426 has a register that reports channel faults such as overcurrent and continuous DC output. Add reporting of this here. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas6424.c | 52 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/tas6424.h | 10 ++++++++ 2 files changed, 56 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aac559fffc1a..36aebdb8f55c 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -41,6 +41,7 @@ struct tas6424_data { struct regmap *regmap; struct regulator_bulk_data supplies[TAS6424_NUM_SUPPLIES]; struct delayed_work fault_check_work; + unsigned int last_cfault; unsigned int last_fault1; unsigned int last_fault2; unsigned int last_warn; @@ -406,6 +407,51 @@ static void tas6424_fault_check_work(struct work_struct *work) unsigned int reg; int ret; + ret = regmap_read(tas6424->regmap, TAS6424_CHANNEL_FAULT, &reg); + if (ret < 0) { + dev_err(dev, "failed to read CHANNEL_FAULT register: %d\n", ret); + goto out; + } + + if (!reg) { + tas6424->last_cfault = reg; + goto check_global_fault1_reg; + } + + /* + * Only flag errors once for a given occurrence. This is needed as + * the TAS6424 will take time clearing the fault condition internally + * during which we don't want to bombard the system with the same + * error message over and over. + */ + if ((reg & TAS6424_FAULT_OC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH1)) + dev_crit(dev, "experienced a channel 1 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH2)) + dev_crit(dev, "experienced a channel 2 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH3)) + dev_crit(dev, "experienced a channel 3 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_OC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_OC_CH4)) + dev_crit(dev, "experienced a channel 4 overcurrent fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH1) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH1)) + dev_crit(dev, "experienced a channel 1 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH2) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH2)) + dev_crit(dev, "experienced a channel 2 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH3) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH3)) + dev_crit(dev, "experienced a channel 3 DC fault\n"); + + if ((reg & TAS6424_FAULT_DC_CH4) && !(tas6424->last_cfault & TAS6424_FAULT_DC_CH4)) + dev_crit(dev, "experienced a channel 4 DC fault\n"); + + /* Store current fault1 value so we can detect any changes next time */ + tas6424->last_cfault = reg; + +check_global_fault1_reg: ret = regmap_read(tas6424->regmap, TAS6424_GLOB_FAULT1, &reg); if (ret < 0) { dev_err(dev, "failed to read GLOB_FAULT1 register: %d\n", ret); @@ -429,12 +475,6 @@ static void tas6424_fault_check_work(struct work_struct *work) goto check_global_fault2_reg; } - /* - * Only flag errors once for a given occurrence. This is needed as - * the TAS6424 will take time clearing the fault condition internally - * during which we don't want to bombard the system with the same - * error message over and over. - */ if ((reg & TAS6424_FAULT_PVDD_OV) && !(tas6424->last_fault1 & TAS6424_FAULT_PVDD_OV)) dev_crit(dev, "experienced a PVDD overvoltage fault\n"); diff --git a/sound/soc/codecs/tas6424.h b/sound/soc/codecs/tas6424.h index b5958c45ed0e..c67a7835ca66 100644 --- a/sound/soc/codecs/tas6424.h +++ b/sound/soc/codecs/tas6424.h @@ -115,6 +115,16 @@ #define TAS6424_LDGBYPASS_SHIFT 0 #define TAS6424_LDGBYPASS_MASK BIT(TAS6424_LDGBYPASS_SHIFT) +/* TAS6424_GLOB_FAULT1_REG */ +#define TAS6424_FAULT_OC_CH1 BIT(7) +#define TAS6424_FAULT_OC_CH2 BIT(6) +#define TAS6424_FAULT_OC_CH3 BIT(5) +#define TAS6424_FAULT_OC_CH4 BIT(4) +#define TAS6424_FAULT_DC_CH1 BIT(3) +#define TAS6424_FAULT_DC_CH2 BIT(2) +#define TAS6424_FAULT_DC_CH3 BIT(1) +#define TAS6424_FAULT_DC_CH4 BIT(0) + /* TAS6424_GLOB_FAULT1_REG */ #define TAS6424_FAULT_CLOCK BIT(4) #define TAS6424_FAULT_PVDD_OV BIT(3) -- 2.19.0.rc1

1 0

[alsa-devel] [PATCH 1/3] ASoC: tas6424: Save last fault register even when clear
by Andrew F. Davis 31 Aug '18

31 Aug '18

When there is no fault bit set in a fault register we skip the fault reporting section for that register. This also skips over saving that registers value. We save the value so we will not double report an error, but if an error clears then returns we will also not report it as we did not save the all cleared register value. Fix this by saving the fault register value in the all clear path. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- sound/soc/codecs/tas6424.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index 14999b999fd3..0d6145549a98 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -424,8 +424,10 @@ static void tas6424_fault_check_work(struct work_struct *work) TAS6424_FAULT_PVDD_UV | TAS6424_FAULT_VBAT_UV; - if (reg) + if (!reg) { + tas6424->last_fault1 = reg; goto check_global_fault2_reg; + } /* * Only flag errors once for a given occurrence. This is needed as @@ -461,8 +463,10 @@ static void tas6424_fault_check_work(struct work_struct *work) TAS6424_FAULT_OTSD_CH3 | TAS6424_FAULT_OTSD_CH4; - if (!reg) + if (!reg) { + tas6424->last_fault2 = reg; goto check_warn_reg; + } if ((reg & TAS6424_FAULT_OTSD) && !(tas6424->last_fault2 & TAS6424_FAULT_OTSD)) dev_crit(dev, "experienced a global overtemp shutdown\n"); @@ -497,8 +501,10 @@ static void tas6424_fault_check_work(struct work_struct *work) TAS6424_WARN_VDD_OTW_CH3 | TAS6424_WARN_VDD_OTW_CH4; - if (!reg) + if (!reg) { + tas6424->last_warn = reg; goto out; + } if ((reg & TAS6424_WARN_VDD_UV) && !(tas6424->last_warn & TAS6424_WARN_VDD_UV)) dev_warn(dev, "experienced a VDD under voltage condition\n"); -- 2.18.0

1 2

[alsa-devel] [RFC] ALSA: Add a static keep_iface quirk for two known devices
by news＠aboehler.at 31 Aug '18

31 Aug '18

From: Andreas Böhler <news(a)aboehler.at> Hi, I own a Phonic Helix Board 12 Universal USB sound mixer that I couldn't get to work properly on my box. After some debugging, it turned out that the playback stream is muted when the capture interface is brought down. Since I was on 4.14, I implemented a quirk there and wanted to port it to 4.18 before submitting it - just to find out that the keep_iface control is very much the same approach. This patch adds a mixer quirk for known devices such as the Phonic Helix Board 12 Universal and the Behringer FCA610 (tested and reported by Takashi Sakamoto). For these devices, the control is not created and keep_iface is simply set to true. Signed-off-by: Andreas Böhler <dev(a)aboehler.at> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c63c84b54969..f11208fb4e58 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3474,9 +3474,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (err < 0) goto _error; - err = create_keep_iface_ctl(mixer); - if (err < 0) - goto _error; + /* For known cards, make the keep_iface quirk static + * otherwiese, add a mixer control */ + if(snd_usb_mixer_keep_iface_static_quirk(mixer)) { + mixer->chip->keep_iface = true; + } else { + err = create_keep_iface_ctl(mixer); + if (err < 0) + goto _error; + } snd_usb_mixer_apply_create_quirk(mixer); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index cbfb48bdea51..414aa1f3a170 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2000,3 +2000,13 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, } } +bool snd_usb_mixer_keep_iface_static_quirk(struct usb_mixer_interface *mixer) +{ + switch (mixer->chip->usb_id) { + case USB_ID(0x170b, 0x0015): /* Phonic Helix Board 12 Universal */ + case USB_ID(0x1397, 0x0003): /* Behringer FCA 610 */ + return true; + } + + return false; +} diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index 52be26db558f..5853e86adffd 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -14,6 +14,8 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl); +bool snd_usb_mixer_keep_iface_static_quirk(struct usb_mixer_interface *mixer); + #ifdef CONFIG_PM void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer); #endif -- 2.18.0

2 1

[alsa-devel] [PATCH 00/15][RFC] ASoC: use snd_soc_dai_link_component style for simple card
by Kuninori Morimoto 31 Aug '18

31 Aug '18

Hi Mark I think ALSA SoC framework needs more update for modern style As 1st step, codec/plaform are updated/converted to component style. I think 2nd step is DAI/dai_link matching. We want to use snd_soc_dai_link_component style instead of legacy style for dai_link matching for CPU/Codec/Platform. like this. [current] struct snd_soc_dai_link { ... *cpu_name; *cpu_of_node; *cpu_dai_name; *codec_name; *codec_of_node; *codec_dai_name; *codecs; num_codecs; *platform_name; *platform_of_node; ... } [we want] struct snd_soc_dai_link { ... *cpus num_cpus; *codecs; num_codecs; *platform; ... } Above sample is still using CPU/Codec/Platform. We can merge it into same snd_soc_dai_link_component list if someone wants it. But anyway, CPU/Codec/Platform style is not so bad, so far. It will be more useful/simple code if we can use snd_soc_dai_link_component for DAI instead of xxx_name / xxx_of_node / xxx_dai_name. Currently, only Codec is supporting/using snd_soc_dai_link_component style. But we want to use it for CPU/Platform, too. I know Shreyas is posting multi CPU style, then, it can support snd_soc_dai_link_component style. Using snd_soc_dai_link_component style for Platform is not difficult I think. [09/15] patch add it on soc-core. But, I can't test for all boards/platforms, thus, I added [RFC] for these patches. Now, we are still missing CPU part, but can use it for Codec/Platform. These patches use it on simple/audio cards for Codec/Platform. If these are OK, and if we could have CPU support in the future, we can convert current legacy style to snd_soc_dai_link_component style for all sound cards. 1) - 8) use snd_soc_dai_link_component for Codec on simple/audio card 9) - 15) use snd_soc_dai_link_component for Platform on simple/audio card Kuninori Morimoto (15): 1) ASoC: simple_card_utils: support snd_soc_dai_link_component style for codec 2) ASoC: simple-card: support snd_soc_dai_link_component style for codec 3) ASoC: simple-scu-card: use simple_dai_props 4) ASoC: simple-scu-card: support snd_soc_dai_link_component style for codec 5) ASoC: audio-graph-card: support snd_soc_dai_link_component style for codec 6) ASoC: audio-graph-scu-card: use simple_dai_props 7) ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for codec 8) ASoC: simple-card-util: remove dai_link compatible code for codec 9) ASoC: soc-core: use snd_soc_dai_link_component for platform 10) ASoC: simple-card-util: support snd_soc_dai_link_component style for platform 11) ASoC: simple-card: support snd_soc_dai_link_component style for platform 12) ASoC: simple-scu-card: support snd_soc_dai_link_component style for platform 13) ASoC: audio-graph-card: support snd_soc_dai_link_component style for platform 14) ASoC: audio-graph-scu-card: support snd_soc_dai_link_component style for platform 15) ASoC: simple-card-util: remove dai_link compatible code for platform include/sound/simple_card_utils.h | 27 ++++++++++------ include/sound/soc.h | 2 ++ sound/soc/generic/audio-graph-card.c | 18 +++++++++-- sound/soc/generic/audio-graph-scu-card.c | 55 +++++++++++++++++++++----------- sound/soc/generic/simple-card-utils.c | 47 ++++++++++++++++++++++++--- sound/soc/generic/simple-card.c | 30 ++++++++++++++--- sound/soc/generic/simple-scu-card.c | 54 ++++++++++++++++++++----------- sound/soc/soc-core.c | 48 ++++++++++++++++++++++++---- 8 files changed, 217 insertions(+), 64 deletions(-) -- 2.7.4

2 18

[alsa-devel] Applied "ALSA: hda: move hda_codec.h to include/sound" to the asoc tree
by Mark Brown 30 Aug '18

30 Aug '18

The patch ALSA: hda: move hda_codec.h to include/sound has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b4af16d617add4f6380d6651473b8efba13ca8ca Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 22 Aug 2018 15:24:57 -0500 Subject: [PATCH] ALSA: hda: move hda_codec.h to include/sound As suggested by Takashi, move this header file to make it easier to include from e.g. the Intel Skylake driver in follow-up patches Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- {sound/pci/hda => include/sound}/hda_codec.h | 0 sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/hda_bind.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- 25 files changed, 24 insertions(+), 24 deletions(-) rename {sound/pci/hda => include/sound}/hda_codec.h (100%) diff --git a/sound/pci/hda/hda_codec.h b/include/sound/hda_codec.h similarity index 100% rename from sound/pci/hda/hda_codec.h rename to include/sound/hda_codec.h diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index b9a6b66aeb0e..df0d636145f8 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -13,7 +13,7 @@ #include <linux/export.h> #include <linux/sort.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index d1a6a9c1329a..f1457c6b3969 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -9,7 +9,7 @@ #ifndef __SOUND_HDA_BEEP_H #define __SOUND_HDA_BEEP_H -#include "hda_codec.h" +#include <sound/hda_codec.h> #define HDA_BEEP_MODE_OFF 0 #define HDA_BEEP_MODE_ON 1 diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d361bb77ca00..2222b47d4ec4 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -11,7 +11,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0a5085537034..ccbb0d12b8cc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -27,7 +27,7 @@ #include <linux/pm.h> #include <linux/pm_runtime.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/asoundef.h> #include <sound/tlv.h> #include <sound/initval.h> diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index a68e75b00ea3..55760e5231e6 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -20,7 +20,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include <sound/hda_register.h> #define AZX_MAX_CODECS HDA_MAX_CODECS diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index ba7fe9b6655c..806b12ed44a2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <asm/unaligned.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" enum eld_versions { diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 579984ecdec3..276150f29cda 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -32,7 +32,7 @@ #include <sound/core.h> #include <sound/jack.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index cc009a4a3d1d..268bba6ec985 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -23,7 +23,7 @@ #include <linux/compat.h> #include <linux/nospec.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..c8fde95e2393 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -63,7 +63,7 @@ #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" #include "hda_intel.h" diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index a33234e04d4f..c499727920e6 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -15,7 +15,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c6b778b2580c..a65740419650 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <sound/core.h> #include <linux/module.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" static int dump_coef = -1; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 6ec79c58d48d..c154b19a0c45 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -14,7 +14,7 @@ #include <linux/string.h> #include <linux/export.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include <sound/hda_hwdep.h> #include <sound/minors.h> diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0621920f7617..4bc5232eac1c 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -35,7 +35,7 @@ #include <sound/core.h> #include <sound/initval.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_controller.h" /* Defines for Nvidia Tegra HDA support */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fd476fb40e1b..ebfd0be885b3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -24,7 +24,7 @@ #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index c2d9ee9cfdc0..21d0f0610913 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -22,7 +22,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0166a3d7cd55..a585d0ec6d77 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -32,7 +32,7 @@ #include <linux/io.h> #include <linux/pci.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a7f91be45194..64fa5a82bb9f 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -23,7 +23,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/tlv.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 1b2195dd2b26..52642ba3e2c0 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -25,7 +25,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cfd4e4f97f8f..5592557fe50e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -27,7 +27,7 @@ #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cb587dce67a9..67099cbb6be2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -41,7 +41,7 @@ #include <sound/hdaudio.h> #include <sound/hda_i915.h> #include <sound/hda_chmap.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d117f00d04d..6f3c8e888c2a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -32,7 +32,7 @@ #include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index f63acb1b965c..c49d25bcd7f2 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -27,7 +27,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" /* si3054 verbs */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 046705b4691a..d16a25a395c9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,7 +32,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/jack.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_beep.h" diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 6b9617aee0e6..9f6f13e25145 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -52,7 +52,7 @@ #include <linux/module.h> #include <sound/core.h> #include <sound/asoundef.h> -#include "hda_codec.h" +#include <sound/hda_codec.h> #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" -- 2.18.0

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix cancel_work_sync() stall from jackpoll work
by Takashi Iwai 30 Aug '18

30 Aug '18

On AMD/ATI controllers, the HD-audio controller driver allows a bus reset upon the error recovery, and its procedure includes the cancellation of pending jack polling work as found in snd_hda_bus_codec_reset(). This works usually fine, but it becomes a problem when the reset happens from the jack poll work itself; then calling cancel_work_sync() from the work being processed tries to wait the finish endlessly. As a workaround, this patch adds the check of current_work() and applies the cancel_work_sync() only when it's not from the jackpoll_work. This doesn't fix the root cause of the reported error below, but at least, it eases the unexpected stall of the whole system. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200937 Cc: <stable(a)vger.kernel.org> Cc: Lukas Wunner <lukas(a)wunner.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0a5085537034..26d348b47867 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3935,7 +3935,8 @@ void snd_hda_bus_reset_codecs(struct hda_bus *bus) list_for_each_codec(codec, bus) { /* FIXME: maybe a better way needed for forced reset */ - cancel_delayed_work_sync(&codec->jackpoll_work); + if (current_work() != &codec->jackpoll_work.work) + cancel_delayed_work_sync(&codec->jackpoll_work); #ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { hda_call_codec_suspend(codec); -- 2.18.0

1 0

[alsa-devel] [PATCH 0/2] ASoC: Add support for MikroElektronika PROTO codec
by Codrin Ciubotariu 30 Aug '18

30 Aug '18

This patchset is based on Florian's initial PROTO driver: https://lkml.org/lkml/2013/5/22/342 My changes made the driver more generic, so that we could connect other I2S CPU DAIs, by using device-tree. Tested with sama5d2 I2S driver. Codrin Ciubotariu (2): ASoC: Add driver for PROTO Audio CODEC (with a WM8731) ASoC: mikroe-proto: dt-bindings: add DT bindings for PROTO board .../bindings/sound/mikroe,mikroe-proto.txt | 23 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/atmel/Kconfig | 7 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mikroe-proto.c | 189 ++++++++++++++++++ 5 files changed, 222 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mikroe,mikroe-proto.txt create mode 100644 sound/soc/atmel/mikroe-proto.c -- 2.17.1

4 9

[alsa-devel] [PATCH] ALSA: hda - Clean up jackpoll_ms option handling
by Takashi Iwai 30 Aug '18

30 Aug '18

Currently the jackpoll_ms option value is passed indirectly by referring to an array in chip->jackpoll_ms although each card needs to see only the assigned value. Also, the sanity check is done at each time in get_jackpoll_interval() although basically jackpoll_ms option is a read-only, hence we need to evaluate only once at probe time. This patch is the code simplification about the above points: the jack polling interval is directly set to chip->jackpoll_interval so that it can be simply copied to each codec. No functional change but only code reduction. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 23 +---------------------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 3 ++- 3 files changed, 4 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 53bdf76fcbba..fe2506672a72 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1223,27 +1223,6 @@ void snd_hda_bus_reset(struct hda_bus *bus) bus->in_reset = 0; } -static int get_jackpoll_interval(struct azx *chip) -{ - int i; - unsigned int j; - - if (!chip->jackpoll_ms) - return 0; - - i = chip->jackpoll_ms[chip->dev_index]; - if (i == 0) - return 0; - if (i < 50 || i > 60000) - j = 0; - else - j = msecs_to_jiffies(i); - if (j == 0) - dev_warn(chip->card->dev, - "jackpoll_ms value out of range: %d\n", i); - return j; -} - /* HD-audio bus initialization */ int azx_bus_init(struct azx *chip, const char *model, const struct hdac_io_ops *io_ops) @@ -1326,7 +1305,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) err = snd_hda_codec_new(&chip->bus, chip->card, c, &codec); if (err < 0) continue; - codec->jackpoll_interval = get_jackpoll_interval(chip); + codec->jackpoll_interval = chip->jackpoll_interval; codec->beep_mode = chip->beep_mode; codecs++; } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 3c9a2f81cfbf..33b99839753d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -121,7 +121,7 @@ struct azx { int capture_streams; int capture_index_offset; int num_streams; - const int *jackpoll_ms; /* per-card jack poll interval */ + int jackpoll_interval; /* jack poll interval in jiffies */ /* Register interaction. */ const struct hda_controller_ops *ops; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a3ce9c60f50..12647de19381 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1676,7 +1676,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->driver_type = driver_caps & 0xff; check_msi(chip); chip->dev_index = dev; - chip->jackpoll_ms = jackpoll_ms; + if (jackpoll_ms[dev] >= 50 && jackpoll_ms[dev] <= 60000) + chip->jackpoll_interval = msecs_to_jiffies(jackpoll_ms[dev]); INIT_LIST_HEAD(&chip->pcm_list); INIT_WORK(&hda->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&hda->list); -- 2.18.0

1 0

[alsa-devel] [PATCH] Patch for sun4i-i2s driver to allow 24-bit audio
by Jonathan Fether 30 Aug '18

30 Aug '18

This patch adds 24-bit sample width support for the sun4i-i2s driver. Jonathan Fether (1): ASoC: sun4i-i2s: Add 24-bit sample width support sound/soc/sunxi/sun4i-i2s.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) -- 1.9.1

1 1

[alsa-devel] Adding DC offset to one channel in alsa config
by Pavel Hofman 29 Aug '18

29 Aug '18

Hi, Please is there a way to add a fixed DC offset to one channel stream in alsa config (.asoundrc)? We are playing with single-ended -> virtual balanced setup ( http://www.diyaudio.com/forums/pc-based/326747-virtual-balanced-regular-sou… ) using the route plugin. The setup works surprisingly nice. Unless one of the input channel AD convertors has a DC offset which ruins the subtracted signal. It is easy to measure the offset. Such as e.g. type route ttable { 0.0= 0.49 + 0.001 0.1= -0.49 } Thanks a lot for any suggestions. Best regards, Pavel.

1 0

[alsa-devel] Applied "ASoC: dmic: add Kconfig prompt for the generic dmic codec." to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: dmic: add Kconfig prompt for the generic dmic codec. has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 26bcf1c368d9460987d597fb0476d60e51a1bf82 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Wed, 29 Aug 2018 17:00:48 +0200 Subject: [PATCH] ASoC: dmic: add Kconfig prompt for the generic dmic codec. Add Kconfig prompt for the generic digital mic to make it configurable through menuconfig Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index adaf26e1989c..9989d35e0fc6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -578,7 +578,11 @@ config SND_SOC_DA9055 tristate config SND_SOC_DMIC - tristate + tristate "Generic Digital Microphone CODEC" + depends on GPIOLIB + help + Enable support for the Generic Digital Microphone CODEC. + Select this if your sound card has DMICs. config SND_SOC_HDMI_CODEC tristate -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: dmic: add DT module alias" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: dmic: add DT module alias has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cb06a037f8362e250a6e61872ffa01ab086ec9e2 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Wed, 29 Aug 2018 17:00:49 +0200 Subject: [PATCH] ASoC: dmic: add DT module alias Before this patch the only alias provided by the dmic module is: alias: platform:dmic-codec Device instantiated from DT will not probe automatically with this After this patch, here is the new alias list: alias: platform:dmic-codec alias: of:N*T*Cdmic-codecC* alias: of:N*T*Cdmic-codec Now the dmic codec probes automatically when instantiated from DT. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/dmic.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 8c4926df9286..71322e0410ee 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -148,6 +148,7 @@ static const struct of_device_id dmic_dev_match[] = { {.compatible = "dmic-codec"}, {} }; +MODULE_DEVICE_TABLE(of, dmic_dev_match); static struct platform_driver dmic_driver = { .driver = { -- 2.18.0

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[alsa-devel] Applied "ASoC: meson: add axg pdm input DT binding documentation" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: meson: add axg pdm input DT binding documentation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5cc3f8f89f2a42992315c4917e0c67ccb0269dec Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Wed, 29 Aug 2018 17:00:50 +0200 Subject: [PATCH] ASoC: meson: add axg pdm input DT binding documentation Add the DT binding documentation for axg's PDM input Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/amlogic,axg-pdm.txt | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt new file mode 100644 index 000000000000..5672d0bc5b16 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt @@ -0,0 +1,24 @@ +* Amlogic Audio PDM input + +Required properties: +- compatible: 'amlogic,axg-pdm' +- reg: physical base address of the controller and length of memory + mapped region. +- clocks: list of clock phandle, one for each entry clock-names. +- clock-names: should contain the following: + * "pclk" : peripheral clock. + * "dclk" : pdm digital clock + * "sysclk" : dsp system clock +- #sound-dai-cells: must be 0. + +Example of PDM on the A113 SoC: + +pdm: audio-controller@ff632000 { + compatible = "amlogic,axg-pdm"; + reg = <0x0 0xff632000 0x0 0x34>; + #sound-dai-cells = <0>; + clocks = <&clkc_audio AUD_CLKID_PDM>, + <&clkc_audio AUD_CLKID_PDM_DCLK>, + <&clkc_audio AUD_CLKID_PDM_SYSCLK>; + clock-names = "pclk", "dclk", "sysclk"; +}; -- 2.18.0

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[alsa-devel] Applied "ASoC: meson: add axg pdm input" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: meson: add axg pdm input has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2cfc123eea7477f26f59506fb45f25cb09ee1591 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Wed, 29 Aug 2018 17:00:51 +0200 Subject: [PATCH] ASoC: meson: add axg pdm input Add pdm input driver for the device found on the amlogic AXG SoC family Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/Kconfig | 9 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-pdm.c | 654 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 665 insertions(+) create mode 100644 sound/soc/meson/axg-pdm.c diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 2ccbadc387de..8b8426ed2363 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -54,6 +54,7 @@ config SND_MESON_AXG_SOUND_CARD imply SND_MESON_AXG_TDMIN imply SND_MESON_AXG_TDMOUT imply SND_MESON_AXG_SPDIFOUT + imply SND_MESON_AXG_PDM help Select Y or M to add support for the AXG SoC sound card @@ -66,4 +67,12 @@ config SND_MESON_AXG_SPDIFOUT Select Y or M to add support for SPDIF output serializer embedded in the Amlogic AXG SoC family +config SND_MESON_AXG_PDM + tristate "Amlogic AXG PDM Input Support" + imply SND_SOC_DMIC + imply COMMON_CLK_AXG_AUDIO + help + Select Y or M to add support for PDM input embedded + in the Amlogic AXG SoC family + endmenu diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index c5e003b093db..4cd25104029d 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -9,6 +9,7 @@ snd-soc-meson-axg-tdmin-objs := axg-tdmin.o snd-soc-meson-axg-tdmout-objs := axg-tdmout.o snd-soc-meson-axg-sound-card-objs := axg-card.o snd-soc-meson-axg-spdifout-objs := axg-spdifout.o +snd-soc-meson-axg-pdm-objs := axg-pdm.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o obj-$(CONFIG_SND_MESON_AXG_FRDDR) += snd-soc-meson-axg-frddr.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_MESON_AXG_TDMIN) += snd-soc-meson-axg-tdmin.o obj-$(CONFIG_SND_MESON_AXG_TDMOUT) += snd-soc-meson-axg-tdmout.o obj-$(CONFIG_SND_MESON_AXG_SOUND_CARD) += snd-soc-meson-axg-sound-card.o obj-$(CONFIG_SND_MESON_AXG_SPDIFOUT) += snd-soc-meson-axg-spdifout.o +obj-$(CONFIG_SND_MESON_AXG_PDM) += snd-soc-meson-axg-pdm.o diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c new file mode 100644 index 000000000000..9d5684493ffc --- /dev/null +++ b/sound/soc/meson/axg-pdm.c @@ -0,0 +1,654 @@ +// SPDX-License-Identifier: (GPL-2.0 OR MIT) +// +// Copyright (c) 2018 BayLibre, SAS. +// Author: Jerome Brunet <jbrunet(a)baylibre.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_irq.h> +#include <linux/of_platform.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/pcm_params.h> + +#define PDM_CTRL 0x00 +#define PDM_CTRL_EN BIT(31) +#define PDM_CTRL_OUT_MODE BIT(29) +#define PDM_CTRL_BYPASS_MODE BIT(28) +#define PDM_CTRL_RST_FIFO BIT(16) +#define PDM_CTRL_CHAN_RSTN_MASK GENMASK(15, 8) +#define PDM_CTRL_CHAN_RSTN(x) ((x) << 8) +#define PDM_CTRL_CHAN_EN_MASK GENMASK(7, 0) +#define PDM_CTRL_CHAN_EN(x) ((x) << 0) +#define PDM_HCIC_CTRL1 0x04 +#define PDM_FILTER_EN BIT(31) +#define PDM_HCIC_CTRL1_GAIN_SFT_MASK GENMASK(29, 24) +#define PDM_HCIC_CTRL1_GAIN_SFT(x) ((x) << 24) +#define PDM_HCIC_CTRL1_GAIN_MULT_MASK GENMASK(23, 16) +#define PDM_HCIC_CTRL1_GAIN_MULT(x) ((x) << 16) +#define PDM_HCIC_CTRL1_DSR_MASK GENMASK(8, 4) +#define PDM_HCIC_CTRL1_DSR(x) ((x) << 4) +#define PDM_HCIC_CTRL1_STAGE_NUM_MASK GENMASK(3, 0) +#define PDM_HCIC_CTRL1_STAGE_NUM(x) ((x) << 0) +#define PDM_HCIC_CTRL2 0x08 +#define PDM_F1_CTRL 0x0c +#define PDM_LPF_ROUND_MODE_MASK GENMASK(17, 16) +#define PDM_LPF_ROUND_MODE(x) ((x) << 16) +#define PDM_LPF_DSR_MASK GENMASK(15, 12) +#define PDM_LPF_DSR(x) ((x) << 12) +#define PDM_LPF_STAGE_NUM_MASK GENMASK(8, 0) +#define PDM_LPF_STAGE_NUM(x) ((x) << 0) +#define PDM_LPF_MAX_STAGE 336 +#define PDM_LPF_NUM 3 +#define PDM_F2_CTRL 0x10 +#define PDM_F3_CTRL 0x14 +#define PDM_HPF_CTRL 0x18 +#define PDM_HPF_SFT_STEPS_MASK GENMASK(20, 16) +#define PDM_HPF_SFT_STEPS(x) ((x) << 16) +#define PDM_HPF_OUT_FACTOR_MASK GENMASK(15, 0) +#define PDM_HPF_OUT_FACTOR(x) ((x) << 0) +#define PDM_CHAN_CTRL 0x1c +#define PDM_CHAN_CTRL_POINTER_WIDTH 8 +#define PDM_CHAN_CTRL_POINTER_MAX ((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1) +#define PDM_CHAN_CTRL_NUM 4 +#define PDM_CHAN_CTRL1 0x20 +#define PDM_COEFF_ADDR 0x24 +#define PDM_COEFF_DATA 0x28 +#define PDM_CLKG_CTRL 0x2c +#define PDM_STS 0x30 + +struct axg_pdm_lpf { + unsigned int ds; + unsigned int round_mode; + const unsigned int *tap; + unsigned int tap_num; +}; + +struct axg_pdm_hcic { + unsigned int shift; + unsigned int mult; + unsigned int steps; + unsigned int ds; +}; + +struct axg_pdm_hpf { + unsigned int out_factor; + unsigned int steps; +}; + +struct axg_pdm_filters { + struct axg_pdm_hcic hcic; + struct axg_pdm_hpf hpf; + struct axg_pdm_lpf lpf[PDM_LPF_NUM]; +}; + +struct axg_pdm_cfg { + const struct axg_pdm_filters *filters; + unsigned int sys_rate; +}; + +struct axg_pdm { + const struct axg_pdm_cfg *cfg; + struct regmap *map; + struct clk *dclk; + struct clk *sysclk; + struct clk *pclk; +}; + +static void axg_pdm_enable(struct regmap *map) +{ + /* Reset AFIFO */ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, PDM_CTRL_RST_FIFO); + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_RST_FIFO, 0); + + /* Enable PDM */ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, PDM_CTRL_EN); +} + +static void axg_pdm_disable(struct regmap *map) +{ + regmap_update_bits(map, PDM_CTRL, PDM_CTRL_EN, 0); +} + +static void axg_pdm_filters_enable(struct regmap *map, bool enable) +{ + unsigned int val = enable ? PDM_FILTER_EN : 0; + + regmap_update_bits(map, PDM_HCIC_CTRL1, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F1_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F2_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_F3_CTRL, PDM_FILTER_EN, val); + regmap_update_bits(map, PDM_HPF_CTRL, PDM_FILTER_EN, val); +} + +static int axg_pdm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + axg_pdm_enable(priv->map); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + axg_pdm_disable(priv->map); + return 0; + + default: + return -EINVAL; + } +} + +static unsigned int axg_pdm_get_os(struct axg_pdm *priv) +{ + const struct axg_pdm_filters *filters = priv->cfg->filters; + unsigned int os = filters->hcic.ds; + int i; + + /* + * The global oversampling factor is defined by the down sampling + * factor applied by each filter (HCIC and LPFs) + */ + + for (i = 0; i < PDM_LPF_NUM; i++) + os *= filters->lpf[i].ds; + + return os; +} + +static int axg_pdm_set_sysclk(struct axg_pdm *priv, unsigned int os, + unsigned int rate) +{ + unsigned int sys_rate = os * 2 * rate * PDM_CHAN_CTRL_POINTER_MAX; + + /* + * Set the default system clock rate unless it is too fast for + * for the requested sample rate. In this case, the sample pointer + * counter could overflow so set a lower system clock rate + */ + if (sys_rate < priv->cfg->sys_rate) + return clk_set_rate(priv->sysclk, sys_rate); + + return clk_set_rate(priv->sysclk, priv->cfg->sys_rate); +} + +static int axg_pdm_set_sample_pointer(struct axg_pdm *priv) +{ + unsigned int spmax, sp, val; + int i; + + /* Max sample counter value per half period of dclk */ + spmax = DIV_ROUND_UP_ULL((u64)clk_get_rate(priv->sysclk), + clk_get_rate(priv->dclk) * 2); + + /* Check if sysclk is not too fast - should not happen */ + if (WARN_ON(spmax > PDM_CHAN_CTRL_POINTER_MAX)) + return -EINVAL; + + /* Capture the data when we are at 75% of the half period */ + sp = spmax * 3 / 4; + + for (i = 0, val = 0; i < PDM_CHAN_CTRL_NUM; i++) + val |= sp << (PDM_CHAN_CTRL_POINTER_WIDTH * i); + + regmap_write(priv->map, PDM_CHAN_CTRL, val); + regmap_write(priv->map, PDM_CHAN_CTRL1, val); + + return 0; +} + +static void axg_pdm_set_channel_mask(struct axg_pdm *priv, + unsigned int channels) +{ + unsigned int mask = GENMASK(channels - 1, 0); + + /* Put all channel in reset */ + regmap_update_bits(priv->map, PDM_CTRL, + PDM_CTRL_CHAN_RSTN_MASK, 0); + + /* Take the necessary channels out of reset and enable them */ + regmap_update_bits(priv->map, PDM_CTRL, + PDM_CTRL_CHAN_RSTN_MASK | + PDM_CTRL_CHAN_EN_MASK, + PDM_CTRL_CHAN_RSTN(mask) | + PDM_CTRL_CHAN_EN(mask)); +} + +static int axg_pdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + unsigned int os = axg_pdm_get_os(priv); + unsigned int rate = params_rate(params); + unsigned int val; + int ret; + + switch (params_width(params)) { + case 24: + val = PDM_CTRL_OUT_MODE; + break; + case 32: + val = 0; + break; + default: + dev_err(dai->dev, "unsupported sample width\n"); + return -EINVAL; + } + + regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_OUT_MODE, val); + + ret = axg_pdm_set_sysclk(priv, os, rate); + if (ret) { + dev_err(dai->dev, "failed to set system clock\n"); + return ret; + } + + ret = clk_set_rate(priv->dclk, rate * os); + if (ret) { + dev_err(dai->dev, "failed to set dclk\n"); + return ret; + } + + ret = axg_pdm_set_sample_pointer(priv); + if (ret) { + dev_err(dai->dev, "invalid clock setting\n"); + return ret; + } + + axg_pdm_set_channel_mask(priv, params_channels(params)); + + return 0; +} + +static int axg_pdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->dclk); + if (ret) { + dev_err(dai->dev, "enabling dclk failed\n"); + return ret; + } + + /* Enable the filters */ + axg_pdm_filters_enable(priv->map, true); + + return ret; +} + +static void axg_pdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + axg_pdm_filters_enable(priv->map, false); + clk_disable_unprepare(priv->dclk); +} + +static const struct snd_soc_dai_ops axg_pdm_dai_ops = { + .trigger = axg_pdm_trigger, + .hw_params = axg_pdm_hw_params, + .startup = axg_pdm_startup, + .shutdown = axg_pdm_shutdown, +}; + +static void axg_pdm_set_hcic_ctrl(struct axg_pdm *priv) +{ + const struct axg_pdm_hcic *hcic = &priv->cfg->filters->hcic; + unsigned int val; + + val = PDM_HCIC_CTRL1_STAGE_NUM(hcic->steps); + val |= PDM_HCIC_CTRL1_DSR(hcic->ds); + val |= PDM_HCIC_CTRL1_GAIN_MULT(hcic->mult); + val |= PDM_HCIC_CTRL1_GAIN_SFT(hcic->shift); + + regmap_update_bits(priv->map, PDM_HCIC_CTRL1, + PDM_HCIC_CTRL1_STAGE_NUM_MASK | + PDM_HCIC_CTRL1_DSR_MASK | + PDM_HCIC_CTRL1_GAIN_MULT_MASK | + PDM_HCIC_CTRL1_GAIN_SFT_MASK, + val); +} + +static void axg_pdm_set_lpf_ctrl(struct axg_pdm *priv, unsigned int index) +{ + const struct axg_pdm_lpf *lpf = &priv->cfg->filters->lpf[index]; + unsigned int offset = index * regmap_get_reg_stride(priv->map) + + PDM_F1_CTRL; + unsigned int val; + + val = PDM_LPF_STAGE_NUM(lpf->tap_num); + val |= PDM_LPF_DSR(lpf->ds); + val |= PDM_LPF_ROUND_MODE(lpf->round_mode); + + regmap_update_bits(priv->map, offset, + PDM_LPF_STAGE_NUM_MASK | + PDM_LPF_DSR_MASK | + PDM_LPF_ROUND_MODE_MASK, + val); +} + +static void axg_pdm_set_hpf_ctrl(struct axg_pdm *priv) +{ + const struct axg_pdm_hpf *hpf = &priv->cfg->filters->hpf; + unsigned int val; + + val = PDM_HPF_OUT_FACTOR(hpf->out_factor); + val |= PDM_HPF_SFT_STEPS(hpf->steps); + + regmap_update_bits(priv->map, PDM_HPF_CTRL, + PDM_HPF_OUT_FACTOR_MASK | + PDM_HPF_SFT_STEPS_MASK, + val); +} + +static int axg_pdm_set_lpf_filters(struct axg_pdm *priv) +{ + const struct axg_pdm_lpf *lpf = priv->cfg->filters->lpf; + unsigned int count = 0; + int i, j; + + for (i = 0; i < PDM_LPF_NUM; i++) + count += lpf[i].tap_num; + + /* Make sure the coeffs fit in the memory */ + if (count >= PDM_LPF_MAX_STAGE) + return -EINVAL; + + /* Set the initial APB bus register address */ + regmap_write(priv->map, PDM_COEFF_ADDR, 0); + + /* Set the tap filter values of all 3 filters */ + for (i = 0; i < PDM_LPF_NUM; i++) { + axg_pdm_set_lpf_ctrl(priv, i); + + for (j = 0; j < lpf[i].tap_num; j++) + regmap_write(priv->map, PDM_COEFF_DATA, lpf[i].tap[j]); + } + + return 0; +} + +static int axg_pdm_dai_probe(struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(priv->pclk); + if (ret) { + dev_err(dai->dev, "enabling pclk failed\n"); + return ret; + } + + /* + * sysclk must be set and enabled as well to access the pdm registers + * Accessing the register w/o it will give a bus error. + */ + ret = clk_set_rate(priv->sysclk, priv->cfg->sys_rate); + if (ret) { + dev_err(dai->dev, "setting sysclk failed\n"); + goto err_pclk; + } + + ret = clk_prepare_enable(priv->sysclk); + if (ret) { + dev_err(dai->dev, "enabling sysclk failed\n"); + goto err_pclk; + } + + /* Make sure the device is initially disabled */ + axg_pdm_disable(priv->map); + + /* Make sure filter bypass is disabled */ + regmap_update_bits(priv->map, PDM_CTRL, PDM_CTRL_BYPASS_MODE, 0); + + /* Load filter settings */ + axg_pdm_set_hcic_ctrl(priv); + axg_pdm_set_hpf_ctrl(priv); + + ret = axg_pdm_set_lpf_filters(priv); + if (ret) { + dev_err(dai->dev, "invalid filter configuration\n"); + goto err_sysclk; + } + + return 0; + +err_sysclk: + clk_disable_unprepare(priv->sysclk); +err_pclk: + clk_disable_unprepare(priv->pclk); + return ret; +} + +static int axg_pdm_dai_remove(struct snd_soc_dai *dai) +{ + struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable_unprepare(priv->sysclk); + clk_disable_unprepare(priv->pclk); + + return 0; +} + +static struct snd_soc_dai_driver axg_pdm_dai_drv = { + .name = "PDM", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 48000, + .formats = (SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &axg_pdm_dai_ops, + .probe = axg_pdm_dai_probe, + .remove = axg_pdm_dai_remove, +}; + +static const struct snd_soc_component_driver axg_pdm_component_drv = {}; + +static const struct regmap_config axg_pdm_regmap_cfg = { + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 4, + .max_register = PDM_STS, +}; + +static const unsigned int lpf1_default_tap[] = { + 0x000014, 0xffffb2, 0xfffed9, 0xfffdce, 0xfffd45, + 0xfffe32, 0x000147, 0x000645, 0x000b86, 0x000e21, + 0x000ae3, 0x000000, 0xffeece, 0xffdca8, 0xffd212, + 0xffd7d1, 0xfff2a7, 0x001f4c, 0x0050c2, 0x0072aa, + 0x006ff1, 0x003c32, 0xffdc4e, 0xff6a18, 0xff0fef, + 0xfefbaf, 0xff4c40, 0x000000, 0x00ebc8, 0x01c077, + 0x02209e, 0x01c1a4, 0x008e60, 0xfebe52, 0xfcd690, + 0xfb8fa5, 0xfba498, 0xfd9812, 0x0181ce, 0x06f5f3, + 0x0d112f, 0x12a958, 0x169686, 0x18000e, 0x169686, + 0x12a958, 0x0d112f, 0x06f5f3, 0x0181ce, 0xfd9812, + 0xfba498, 0xfb8fa5, 0xfcd690, 0xfebe52, 0x008e60, + 0x01c1a4, 0x02209e, 0x01c077, 0x00ebc8, 0x000000, + 0xff4c40, 0xfefbaf, 0xff0fef, 0xff6a18, 0xffdc4e, + 0x003c32, 0x006ff1, 0x0072aa, 0x0050c2, 0x001f4c, + 0xfff2a7, 0xffd7d1, 0xffd212, 0xffdca8, 0xffeece, + 0x000000, 0x000ae3, 0x000e21, 0x000b86, 0x000645, + 0x000147, 0xfffe32, 0xfffd45, 0xfffdce, 0xfffed9, + 0xffffb2, 0x000014, +}; + +static const unsigned int lpf2_default_tap[] = { + 0x00050a, 0xfff004, 0x0002c1, 0x003c12, 0xffa818, + 0xffc87d, 0x010aef, 0xff5223, 0xfebd93, 0x028f41, + 0xff5c0e, 0xfc63f8, 0x055f81, 0x000000, 0xf478a0, + 0x11c5e3, 0x2ea74d, 0x11c5e3, 0xf478a0, 0x000000, + 0x055f81, 0xfc63f8, 0xff5c0e, 0x028f41, 0xfebd93, + 0xff5223, 0x010aef, 0xffc87d, 0xffa818, 0x003c12, + 0x0002c1, 0xfff004, 0x00050a, +}; + +static const unsigned int lpf3_default_tap[] = { + 0x000000, 0x000081, 0x000000, 0xfffedb, 0x000000, + 0x00022d, 0x000000, 0xfffc46, 0x000000, 0x0005f7, + 0x000000, 0xfff6eb, 0x000000, 0x000d4e, 0x000000, + 0xffed1e, 0x000000, 0x001a1c, 0x000000, 0xffdcb0, + 0x000000, 0x002ede, 0x000000, 0xffc2d1, 0x000000, + 0x004ebe, 0x000000, 0xff9beb, 0x000000, 0x007dd7, + 0x000000, 0xff633a, 0x000000, 0x00c1d2, 0x000000, + 0xff11d5, 0x000000, 0x012368, 0x000000, 0xfe9c45, + 0x000000, 0x01b252, 0x000000, 0xfdebf6, 0x000000, + 0x0290b8, 0x000000, 0xfcca0d, 0x000000, 0x041d7c, + 0x000000, 0xfa8152, 0x000000, 0x07e9c6, 0x000000, + 0xf28fb5, 0x000000, 0x28b216, 0x3fffde, 0x28b216, + 0x000000, 0xf28fb5, 0x000000, 0x07e9c6, 0x000000, + 0xfa8152, 0x000000, 0x041d7c, 0x000000, 0xfcca0d, + 0x000000, 0x0290b8, 0x000000, 0xfdebf6, 0x000000, + 0x01b252, 0x000000, 0xfe9c45, 0x000000, 0x012368, + 0x000000, 0xff11d5, 0x000000, 0x00c1d2, 0x000000, + 0xff633a, 0x000000, 0x007dd7, 0x000000, 0xff9beb, + 0x000000, 0x004ebe, 0x000000, 0xffc2d1, 0x000000, + 0x002ede, 0x000000, 0xffdcb0, 0x000000, 0x001a1c, + 0x000000, 0xffed1e, 0x000000, 0x000d4e, 0x000000, + 0xfff6eb, 0x000000, 0x0005f7, 0x000000, 0xfffc46, + 0x000000, 0x00022d, 0x000000, 0xfffedb, 0x000000, + 0x000081, 0x000000, +}; + +/* + * These values are sane defaults for the axg platform: + * - OS = 64 + * - Latency = 38700 (?) + * + * TODO: There is a lot of different HCIC, LPFs and HPF configurations possible. + * the configuration may depend on the dmic used by the platform, the + * expected tradeoff between latency and quality, etc ... If/When other + * settings are required, we should add a fw interface to this driver to + * load new filter settings. + */ +static const struct axg_pdm_filters axg_default_filters = { + .hcic = { + .shift = 0x15, + .mult = 0x80, + .steps = 7, + .ds = 8, + }, + .hpf = { + .out_factor = 0x8000, + .steps = 13, + }, + .lpf = { + [0] = { + .ds = 2, + .round_mode = 1, + .tap = lpf1_default_tap, + .tap_num = ARRAY_SIZE(lpf1_default_tap), + }, + [1] = { + .ds = 2, + .round_mode = 0, + .tap = lpf2_default_tap, + .tap_num = ARRAY_SIZE(lpf2_default_tap), + }, + [2] = { + .ds = 2, + .round_mode = 1, + .tap = lpf3_default_tap, + .tap_num = ARRAY_SIZE(lpf3_default_tap) + }, + }, +}; + +static const struct axg_pdm_cfg axg_pdm_config = { + .filters = &axg_default_filters, + .sys_rate = 250000000, +}; + +static const struct of_device_id axg_pdm_of_match[] = { + { + .compatible = "amlogic,axg-pdm", + .data = &axg_pdm_config, + }, {} +}; +MODULE_DEVICE_TABLE(of, axg_pdm_of_match); + +static int axg_pdm_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct axg_pdm *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + platform_set_drvdata(pdev, priv); + + priv->cfg = of_device_get_match_data(dev); + if (!priv->cfg) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->map = devm_regmap_init_mmio(dev, regs, &axg_pdm_regmap_cfg); + if (IS_ERR(priv->map)) { + dev_err(dev, "failed to init regmap: %ld\n", + PTR_ERR(priv->map)); + return PTR_ERR(priv->map); + } + + priv->pclk = devm_clk_get(dev, "pclk"); + if (IS_ERR(priv->pclk)) { + ret = PTR_ERR(priv->pclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get pclk: %d\n", ret); + return ret; + } + + priv->dclk = devm_clk_get(dev, "dclk"); + if (IS_ERR(priv->dclk)) { + ret = PTR_ERR(priv->dclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get dclk: %d\n", ret); + return ret; + } + + priv->sysclk = devm_clk_get(dev, "sysclk"); + if (IS_ERR(priv->sysclk)) { + ret = PTR_ERR(priv->sysclk); + if (ret != -EPROBE_DEFER) + dev_err(dev, "failed to get dclk: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(dev, &axg_pdm_component_drv, + &axg_pdm_dai_drv, 1); +} + +static struct platform_driver axg_pdm_pdrv = { + .probe = axg_pdm_probe, + .driver = { + .name = "axg-pdm", + .of_match_table = axg_pdm_of_match, + }, +}; +module_platform_driver(axg_pdm_pdrv); + +MODULE_DESCRIPTION("Amlogic AXG PDM Input driver"); +MODULE_AUTHOR("Jerome Brunet <jbrunet(a)baylibre.com>"); +MODULE_LICENSE("GPL v2"); -- 2.18.0

1 0

[alsa-devel] [PATCH 0/4] ASoC: meson-axg: add pdm input support
by Jerome Brunet 29 Aug '18

29 Aug '18

The purpose of this patchset is to add the support for PDM input backend found on the AXG SoC family. The first 2 patches, for the dmic, are not specific to this amlogic driver but it is useful to have them since the PDM input will use this dmic codec. Jerome Brunet (4): ASoC: dmic: add Kconfig prompt for the generic dmic codec. ASoC: dmic: add DT module alias ASoC: meson: add axg pdm input DT binding documentation ASoC: meson: add axg pdm input .../bindings/sound/amlogic,axg-pdm.txt | 24 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/dmic.c | 1 + sound/soc/meson/Kconfig | 9 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-pdm.c | 654 ++++++++++++++++++ 6 files changed, 695 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 sound/soc/meson/axg-pdm.c -- 2.17.1

1 4

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dba508b5ab1d138bd7543a3f503492f1a173aa32 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index c0bc429249fa..3959e6ad113d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -67,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -320,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -331,7 +334,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static bool adau17x1_has_safeload(struct adau *adau) { @@ -854,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -898,7 +900,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] [PATCH 0/1] ASoC: pcm3060: Add codec driver
by Kirill Marinushkin 29 Aug '18

29 Aug '18

Hello Mark, I am developing sound support for a Linux-based device, with playback and capture through the TI PCM3060 codec. With the following patch, I propose to add the PCM3060 codec driver into the kernel. Best Regards, Kirill Kirill Marinushkin (1): ASoC: pcm3060: Add codec driver .../devicetree/bindings/sound/pcm3060.txt | 17 ++ MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 17 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3060-i2c.c | 61 +++++ sound/soc/codecs/pcm3060-spi.c | 60 +++++ sound/soc/codecs/pcm3060.c | 290 +++++++++++++++++++++ sound/soc/codecs/pcm3060.h | 88 +++++++ 8 files changed, 546 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm3060.txt create mode 100644 sound/soc/codecs/pcm3060-i2c.c create mode 100644 sound/soc/codecs/pcm3060-spi.c create mode 100644 sound/soc/codecs/pcm3060.c create mode 100644 sound/soc/codecs/pcm3060.h -- 2.13.6

2 8

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] [PATCH 1/3] ASoC: cs4265: SOC_SINGLE register value error fix
by Matt Flax 29 Aug '18

29 Aug '18

The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..15b4ae04 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.17.1

3 13

[alsa-devel] Applied "ASoC: max98373: Added speaker FS gain cotnrol register to volatile." to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: max98373: Added speaker FS gain cotnrol register to volatile. has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0d22825255f25adb6a609f130b42c752d3fd0f5d Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Wed, 15 Aug 2018 18:53:38 -0700 Subject: [PATCH] ASoC: max98373: Added speaker FS gain cotnrol register to volatile. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 92b7125ea169..2764fae69333 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -520,6 +520,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98373_R2000_SW_RESET ... MAX98373_R2009_INT_FLAG3: + case MAX98373_R203E_AMP_PATH_GAIN: case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: rt5514: Fix the issue of the delay volume applied again" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: rt5514: Fix the issue of the delay volume applied again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6f0a256253f48095ba2e5bcdfbed41f21643c105 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 15 Aug 2018 14:47:49 +0800 Subject: [PATCH] ASoC: rt5514: Fix the issue of the delay volume applied again After our evaluation, we need to modify the default values to make sure the volume applied immediately. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index dca82dd6e3bf..32fe76c3134a 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: Convert to using %pOFn instead of device_node.name" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: Convert to using %pOFn instead of device_node.name has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5d585e1e756838d91144c3173323b96f5aa12874 Mon Sep 17 00:00:00 2001 From: Rob Herring <robh(a)kernel.org> Date: Tue, 28 Aug 2018 10:44:28 -0500 Subject: [PATCH] ASoC: Convert to using %pOFn instead of device_node.name In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Signed-off-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 +++---- 5 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c1d1d06783e5..57b484768a58 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); + snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b52223..9981668ab590 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, of_node_put(dma_channel_np); return ret; } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn", + (unsigned long long) res.start, dma_channel_np); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2914ba0d965b..b76a5f4f1785 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -478,7 +478,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, ret = axg_card_set_link_name(card, link, "be"); if (ret) - dev_err(card->dev, "error setting %s link name\n", np->name); + dev_err(card->dev, "error setting %pOFn link name\n", np); return ret; } diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index f22654253c43..d597eba61992 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!pdev) { dev_err(&sai_client->pdev->dev, - "Device not found for node %s\n", np_provider->name); + "Device not found for node %pOFn\n", np_provider); return -ENODEV; } diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 06fba9650ac4..56a227e0bd71 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1124,16 +1124,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sync = SAI_SYNC_NONE; if (args.np) { if (args.np == np) { - dev_err(&pdev->dev, "%s sync own reference\n", - np->name); + dev_err(&pdev->dev, "%pOFn sync own reference\n", np); of_node_put(args.np); return -EINVAL; } sai->np_sync_provider = of_get_parent(args.np); if (!sai->np_sync_provider) { - dev_err(&pdev->dev, "%s parent node not found\n", - np->name); + dev_err(&pdev->dev, "%pOFn parent node not found\n", + np); of_node_put(args.np); return -ENODEV; } -- 2.18.0

1 0

[alsa-devel] [PATCH 1/7] ASoC: dapm: Remove clock framework ifdefs
by Charles Keepax 28 Aug '18

28 Aug '18

The clock code now has stub functions defined in its header files so the ifdefs around clocking code should no longer be necessary. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..78ab6965af55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1320,14 +1320,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, soc_dapm_async_complete(w->dapm); -#ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); } else { clk_disable_unprepare(w->clk); return 0; } -#endif + return 0; } EXPORT_SYMBOL_GPL(dapm_clock_event); @@ -3498,7 +3497,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } break; case snd_soc_dapm_clock_supply: -#ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); @@ -3508,9 +3506,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } -#else - return NULL; -#endif break; default: break; -- 2.11.0

2 9

[alsa-devel] [PATCH] ASoC: dpcm: Properly initialise hw->rate_max
by Charles Keepax 28 Aug '18

28 Aug '18

If the CPU DAI does not initialise rate_max, say if using using KNOT or CONTINUOUS, then the rate_max field will be initialised to 0. A value of zero in the rate_max field of the hardware runtime will cause the sound card to support no sample rates at all. Obviously this is not desired, just a different mechanism is being used to apply the constraints. As such update the setting of rate_max in dpcm_init_runtime_hw to be consistent with the non-DPCM cases and set rate_max to UINT_MAX if nothing is defined on the CPU DAI. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e8b98bfd4cf1..eb6f4f1b65a9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1680,7 +1680,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *stream) { runtime->hw.rate_min = stream->rate_min; - runtime->hw.rate_max = stream->rate_max; + runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX); runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: meson: imply clock and reset controllers
by Jerome Brunet 28 Aug '18

28 Aug '18

Add audio clock controller and ARB reset controller module implication for the device using them Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 8af8bc358a90..2ccbadc387de 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -4,6 +4,8 @@ menu "ASoC support for Amlogic platforms" config SND_MESON_AXG_FIFO tristate select REGMAP_MMIO + imply COMMON_CLK_AXG_AUDIO + imply RESET_MESON_AUDIO_ARB config SND_MESON_AXG_FRDDR tristate "Amlogic AXG Playback FIFO support" @@ -22,6 +24,7 @@ config SND_MESON_AXG_TODDR config SND_MESON_AXG_TDM_FORMATTER tristate select REGMAP_MMIO + imply COMMON_CLK_AXG_AUDIO config SND_MESON_AXG_TDM_INTERFACE tristate @@ -58,6 +61,7 @@ config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" select SND_PCM_IEC958 imply SND_SOC_SPDIF + imply COMMON_CLK_AXG_AUDIO help Select Y or M to add support for SPDIF output serializer embedded in the Amlogic AXG SoC family -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: meson: axg-fifo: report interrupt request failure
by Jerome Brunet 28 Aug '18

28 Aug '18

Return value of request_irq() was irgnored. Fix this and report the failure if any Fixes: 6dc4fa179fb8 ("ASoC: meson: add axg fifo base driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 30262550e37b..0e4f65e654c4 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -203,6 +203,8 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) ret = request_irq(fifo->irq, axg_fifo_pcm_irq_block, 0, dev_name(dev), ss); + if (ret) + return ret; /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: meson: axg-tdm: restrict formats depending on slot width
by Jerome Brunet 28 Aug '18

28 Aug '18

Restrict the formats possible on the TDM interface depending on the width of the TDM slot and let dpcm merging do the rest. Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 50 +++++++++++++++++------------ 1 file changed, 29 insertions(+), 21 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 7b8baf46d968..585ce030b79b 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -42,6 +42,7 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, struct axg_tdm_stream *rx = (struct axg_tdm_stream *) dai->capture_dma_data; unsigned int tx_slots, rx_slots; + unsigned int fmt = 0; tx_slots = axg_tdm_slots_total(tx_mask); rx_slots = axg_tdm_slots_total(rx_mask); @@ -52,38 +53,45 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, return -EINVAL; } - /* - * Amend the dai driver channel number and let dpcm channel merge do - * its job - */ - if (tx) { - tx->mask = tx_mask; - dai->driver->playback.channels_max = tx_slots; - } - - if (rx) { - rx->mask = rx_mask; - dai->driver->capture.channels_max = rx_slots; - } - iface->slots = slots; switch (slot_width) { case 0: - /* defaults width to 32 if not provided */ - iface->slot_width = 32; - break; - case 8: - case 16: - case 24: + slot_width = 32; + /* Fall-through */ case 32: - iface->slot_width = slot_width; + fmt |= SNDRV_PCM_FMTBIT_S32_LE; + /* Fall-through */ + case 24: + fmt |= SNDRV_PCM_FMTBIT_S24_LE; + fmt |= SNDRV_PCM_FMTBIT_S20_LE; + /* Fall-through */ + case 16: + fmt |= SNDRV_PCM_FMTBIT_S16_LE; + /* Fall-through */ + case 8: + fmt |= SNDRV_PCM_FMTBIT_S8; break; default: dev_err(dai->dev, "unsupported slot width: %d\n", slot_width); return -EINVAL; } + iface->slot_width = slot_width; + + /* Amend the dai driver and let dpcm merge do its job */ + if (tx) { + tx->mask = tx_mask; + dai->driver->playback.channels_max = tx_slots; + dai->driver->playback.formats = fmt; + } + + if (rx) { + rx->mask = rx_mask; + dai->driver->capture.channels_max = rx_slots; + dai->driver->capture.formats = fmt; + } + return 0; } EXPORT_SYMBOL_GPL(axg_tdm_set_tdm_slots); -- 2.17.1

2 1

[alsa-devel] [v2] ASoC: max98373: Added 10ms sleep after amp software reset
by Ryan Lee 28 Aug '18

28 Aug '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes since v1 : Applied usleep_range intead of using mdelay Changes : Applied 10ms delay after amp software reset. 10ms guard time is required for stability. sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 2764fae..6aaff03 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -730,6 +730,7 @@ static int max98373_probe(struct snd_soc_component *component) /* Software Reset */ regmap_write(max98373->regmap, MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); + usleep_range(10000, 11000); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,6 +819,7 @@ static int max98373_resume(struct device *dev) regmap_write(max98373->regmap, MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); + usleep_range(10000, 11000); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: rt5682: Change DAC/ADC volume scale" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: rt5682: Change DAC/ADC volume scale has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7509487785d7a2bf3606cf26710f0ca29e9ca94d Mon Sep 17 00:00:00 2001 From: Shuming Fan <shumingf(a)realtek.com> Date: Fri, 24 Aug 2018 10:52:19 +0800 Subject: [PATCH] ASoC: rt5682: Change DAC/ADC volume scale The step of DAC/ADC volume scale changes from 0.375dB to 0.75dB Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/rt5682.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 640d400ca013..afe7d5b19313 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -750,8 +750,8 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ @@ -1114,7 +1114,7 @@ static const struct snd_kcontrol_new rt5682_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, - RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), /* IN Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, @@ -1124,7 +1124,7 @@ static const struct snd_kcontrol_new rt5682_snd_controls[] = { SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, - RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, -- 2.18.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5682: Update calibration function
by shumingf＠realtek.com 28 Aug '18

28 Aug '18

From: Shuming Fan <shumingf(a)realtek.com> New calibration sequence allows rt5682 do calibration without MCLK. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 640d400..0ea2759 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2454,27 +2454,15 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2bf); usleep_range(15000, 20000); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2bf); - regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); - regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8001); - regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); - regmap_write(rt5682->regmap, RT5682_STO1_DAC_MIXER, 0x2080); - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x4040); - regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0069); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0300); + regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x8000); + regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_HP_CTRL_2, 0x6000); - regmap_write(rt5682->regmap, RT5682_HP_CHARGE_PUMP_1, 0x0f26); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7f05); - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); - regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); - regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_9, 0x000f); - regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x8d01); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); regmap_write(rt5682->regmap, RT5682_HP_LOGIC_CTRL_2, 0x0004); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_3, 0x06a1); - regmap_write(rt5682->regmap, RT5682_A_DAC1_MUX, 0x0311); - regmap_write(rt5682->regmap, RT5682_RESET_HPF_CTRL, 0x0000); - regmap_write(rt5682->regmap, RT5682_ADC_STO1_HP_CTRL_1, 0x3320); + regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0x7c00); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_1, 0xfc00); @@ -2490,7 +2478,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) pr_err("HP Calibration Failure\n"); /* restore settings */ - regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); mutex_unlock(&rt5682->calibrate_mutex); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: wm8804: Add ACPI support
by Pierre-Louis Bossart 28 Aug '18

28 Aug '18

HID made of either Wolfson/CirrusLogic PCI ID + 8804 identifier. This helps enumerate the HifiBerry Digi+ HAT boards on the Up2 platform. The scripts at https://github.com/thesofproject/acpi-scripts can be used to add the ACPI initrd overlays. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/wm8804-i2c.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c index f27464c2c5ba..79541960f45d 100644 --- a/sound/soc/codecs/wm8804-i2c.c +++ b/sound/soc/codecs/wm8804-i2c.c @@ -13,6 +13,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/acpi.h> #include "wm8804.h" @@ -40,17 +41,29 @@ static const struct i2c_device_id wm8804_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); +#if defined(CONFIG_OF) static const struct of_device_id wm8804_of_match[] = { { .compatible = "wlf,wm8804", }, { } }; MODULE_DEVICE_TABLE(of, wm8804_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id wm8804_acpi_match[] = { + { "1AEC8804", 0 }, /* Wolfson PCI ID + part ID */ + { "10138804", 0 }, /* Cirrus Logic PCI ID + part ID */ + { }, +}; +MODULE_DEVICE_TABLE(acpi, wm8804_acpi_match); +#endif static struct i2c_driver wm8804_i2c_driver = { .driver = { .name = "wm8804", .pm = &wm8804_pm, - .of_match_table = wm8804_of_match, + .of_match_table = of_match_ptr(wm8804_of_match), + .acpi_match_table = ACPI_PTR(wm8804_acpi_match), }, .probe = wm8804_i2c_probe, .remove = wm8804_i2c_remove, -- 2.17.1

3 2

[alsa-devel] Applied "ASoC: Intel: common: add table for HDA-based platforms" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: Intel: common: add table for HDA-based platforms has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8dce1d026da4588382ed8c03e791c7c9b37b22e8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 22 Aug 2018 15:24:58 -0500 Subject: [PATCH] ASoC: Intel: common: add table for HDA-based platforms Expose a table containing machine driver information for HDAudio-based platforms handled by ASoC on Intel hardware. We only set constant values that are valid across multiple platforms. The firmware name used by the DSP will be set dynamically for each platform. The table is made of a single entry for now, if we need more complicated set-up where HDAudio is mixed with ACPI-enumerated devices (I2C, SoundWire) then we'd expect the differentiation to be handled through information provided by the BIOS (as done for KBL Chromebooks). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-acpi-intel-match.h | 6 +++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-hda-match.c | 40 +++++++++++++++++++ 3 files changed, 48 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index bb1d24b703fb..f48f59e5b7b0 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -25,4 +25,10 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_bxt_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[]; +/* + * generic table used for HDA codec-based platforms, possibly with + * additional ACPI-enumerated codecs + */ +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[]; + #endif diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 915a34cdc8ac..c1f50a079d34 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -7,7 +7,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-hsw-bdw-match.o \ soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ - soc-acpi-intel-cnl-match.o + soc-acpi-intel-cnl-match.o \ + soc-acpi-intel-hda-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c new file mode 100644 index 000000000000..533c1064f84b --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c @@ -0,0 +1,40 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Intel Corporation. + +/* + * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration. + * + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "../skylake/skl.h" + +static struct skl_machine_pdata hda_pdata = { + .use_tplg_pcm = true, +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = { + { + /* .id is not used in this file */ + .drv_name = "skl_hda_dsp_generic", + + /* .fw_filename is dynamically set in skylake driver */ + + /* .sof_fw_filename is dynamically set in sof/intel driver */ + + .sof_tplg_filename = "intel/sof-hda-generic.tplg", + + /* + * .machine_quirk and .quirk_data are not used here but + * can be used if we need a more complicated machine driver + * combining HDA+other device (e.g. DMIC). + */ + .pdata = &hda_pdata, + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_hda_machines); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Common ACPI Match module"); -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7c33b5f16915a7bc3d3b81a9a041bdc562f71dfb Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Date: Wed, 22 Aug 2018 15:24:59 -0500 Subject: [PATCH] ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs Add machine driver for Intel platforms (SKL/KBL/BXT/APL) with HDA and iDisp codecs. This patch adds support for only iDisp (HDMI/DP) codec. In the following patches support for HDA codecs will be added. This should work for other Intel platforms as well e.g. GLK,CNL however this series is not tested on all the platforms. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 8 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 103 +++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 +++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 144 +++++++++++++++++++ sound/soc/intel/skylake/skl.h | 2 + 6 files changed, 297 insertions(+) create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index cccda87f4b34..0f0d57859555 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -279,6 +279,14 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. + +config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH + tristate "SKL/KBL/BXT/APL with HDA Codecs" + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC machine driver for Intel platforms + SKL/KBL/BXT/APL with iDisp, HDA audio codecs. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 87ef8b4058e5..6e88373cbe35 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -20,6 +20,7 @@ snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o +snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o @@ -46,3 +47,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c new file mode 100644 index 000000000000..f9917e0f2ba8 --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -0,0 +1,103 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * Common functions used in different Intel machine drivers + */ +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "skl_hda_dsp_common.h" + +#define NAME_SIZE 32 + +int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hda_hdmi_pcm *pcm; + char dai_name[NAME_SIZE]; + + pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d", + ctx->dai_index); + pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name); + if (!pcm->codec_dai) + return -EINVAL; + + pcm->device = device; + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +/* skl_hda_digital audio interface glue - connects codec <--> CPU */ +struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { + /* Back End DAI links */ + { + .name = "iDisp1", + .id = 1, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 2, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 3, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +int skl_hda_hdmi_jack_init(struct snd_soc_card *card) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + struct skl_hda_hdmi_pcm *pcm; + char jack_name[NAME_SIZE]; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &pcm->hdmi_jack, + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &pcm->hdmi_jack); + if (err < 0) + return err; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h new file mode 100644 index 000000000000..b6c79696bfba --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -0,0 +1,38 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2015-18 Intel Corporation. + */ + +/* + * This file defines data structures used in Machine Driver for Intel + * platforms with HDA Codecs. + */ + +#ifndef __SOUND_SOC_HDA_DSP_COMMON_H +#define __SOUND_SOC_HDA_DSP_COMMON_H +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> + +#define HDA_DSP_MAX_BE_DAI_LINKS 3 + +struct skl_hda_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + struct snd_soc_jack hdmi_jack; + int device; +}; + +struct skl_hda_private { + struct list_head hdmi_pcm_list; + int pcm_count; + int dai_index; + const char *platform_name; +}; + +extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; +int skl_hda_hdmi_jack_init(struct snd_soc_card *card); +int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device); + +#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */ diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c new file mode 100644 index 000000000000..920bc2ce22aa --- /dev/null +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -0,0 +1,144 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2015-18 Intel Corporation. + +/* + * Machine Driver for SKL+ platforms with DSP and iDisp, HDA Codecs + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/hdac_hdmi.h" +#include "../skylake/skl.h" +#include "skl_hda_dsp_common.h" + +static const struct snd_soc_dapm_route skl_hda_map[] = { + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, +}; + +static int skl_hda_card_late_probe(struct snd_soc_card *card) +{ + return skl_hda_hdmi_jack_init(card); +} + +static int +skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); + int ret = 0; + + dev_dbg(card->dev, "%s: dai link name - %s\n", __func__, link->name); + link->platform_name = ctx->platform_name; + link->nonatomic = 1; + + if (strstr(link->name, "HDMI")) { + ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count); + + if (ret < 0) + return ret; + + ctx->dai_index++; + } + + ctx->pcm_count++; + return ret; +} + +static struct snd_soc_card hda_soc_card = { + .name = "skl_hda_card", + .owner = THIS_MODULE, + .dai_link = skl_hda_be_dai_links, + .dapm_routes = skl_hda_map, + .add_dai_link = skl_hda_add_dai_link, + .fully_routed = true, + .late_probe = skl_hda_card_late_probe, +}; + +#define IDISP_DAI_COUNT 3 +/* there are two routes per iDisp output */ +#define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) +#define IDISP_CODEC_MASK 0x4 + +static int skl_hda_fill_card_info(struct skl_machine_pdata *pdata) +{ + struct snd_soc_card *card = &hda_soc_card; + u32 codec_count, codec_mask; + int i, num_links, num_route; + + codec_mask = pdata->codec_mask; + codec_count = hweight_long(codec_mask); + + if (codec_count == 1 && pdata->codec_mask & IDISP_CODEC_MASK) { + num_links = IDISP_DAI_COUNT; + num_route = IDISP_ROUTE_COUNT; + } else { + return -EINVAL; + } + + card->num_links = num_links; + card->num_dapm_routes = num_route; + + for (i = 0; i < num_links; i++) + skl_hda_be_dai_links[i].platform_name = pdata->platform; + + return 0; +} + +static int skl_hda_audio_probe(struct platform_device *pdev) +{ + struct skl_machine_pdata *pdata; + struct skl_hda_private *ctx; + int ret; + + dev_dbg(&pdev->dev, "%s: entry\n", __func__); + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + pdata = dev_get_drvdata(&pdev->dev); + if (!pdata) + return -EINVAL; + + ret = skl_hda_fill_card_info(pdata); + if (ret < 0) { + dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); + return ret; + } + + ctx->pcm_count = hda_soc_card.num_links; + ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ + ctx->platform_name = pdata->platform; + + hda_soc_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&hda_soc_card, ctx); + + return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card); +} + +static struct platform_driver skl_hda_audio = { + .probe = skl_hda_audio_probe, + .driver = { + .name = "skl_hda_dsp_generic", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(skl_hda_audio) + +/* Module information */ +MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver"); +MODULE_AUTHOR("Rakesh Ughreja <rakesh.a.ughreja(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_hda_dsp_generic"); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 78aa8bdcb619..4105a9371b64 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -117,6 +117,8 @@ struct skl_dma_params { struct skl_machine_pdata { u32 dmic_num; bool use_tplg_pcm; /* use dais and dai links from topology */ + const char *platform; + u32 codec_mask; }; struct skl_dsp_ops { -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9cdae4352cba3f66d39a4ef78bb726940ae1e513 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Date: Wed, 22 Aug 2018 15:25:00 -0500 Subject: [PATCH] ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails When no I2S based codec entries are found in the BIOS, check if there are any HDA codecs detected on the bus. Based on the number of codecs found take appropriate action in machine driver. If there are two HDA codecs i.e. iDisp + HDA found on the bus, register DAIs and DAI links for both. If only one codec i.e. iDisp is found then load only iDisp machine driver. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 38 ++++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cf09721ca13e..3b836125d1de 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -472,6 +472,25 @@ static struct skl_ssp_clk skl_ssp_clks[] = { {.name = "ssp5_sclkfs"}, }; +static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl *skl, + struct snd_soc_acpi_mach *machines) +{ + struct hdac_bus *bus = skl_to_bus(skl); + struct snd_soc_acpi_mach *mach; + + /* check if we have any codecs detected on bus */ + if (bus->codec_mask == 0) + return NULL; + + /* point to common table */ + mach = snd_soc_acpi_intel_hda_machines; + + /* all entries in the machine table use the same firmware */ + mach->fw_filename = machines->fw_filename; + + return mach; +} + static int skl_find_machine(struct skl *skl, void *driver_data) { struct hdac_bus *bus = skl_to_bus(skl); @@ -479,9 +498,13 @@ static int skl_find_machine(struct skl *skl, void *driver_data) struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); - if (mach == NULL) { - dev_err(bus->dev, "No matching machine driver found\n"); - return -ENODEV; + if (!mach) { + dev_dbg(bus->dev, "No matching I2S machine driver found\n"); + mach = skl_find_hda_machine(skl, driver_data); + if (!mach) { + dev_err(bus->dev, "No matching machine driver found\n"); + return -ENODEV; + } } skl->mach = mach; @@ -498,8 +521,9 @@ static int skl_find_machine(struct skl *skl, void *driver_data) static int skl_machine_device_register(struct skl *skl) { - struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach = skl->mach; + struct hdac_bus *bus = skl_to_bus(skl); + struct skl_machine_pdata *pdata; struct platform_device *pdev; int ret; @@ -516,8 +540,12 @@ static int skl_machine_device_register(struct skl *skl) return -EIO; } - if (mach->pdata) + if (mach->pdata) { + pdata = (struct skl_machine_pdata *)mach->pdata; + pdata->platform = dev_name(bus->dev); + pdata->codec_mask = bus->codec_mask; dev_set_drvdata(&pdev->dev, mach->pdata); + } skl->i2s_dev = pdev; -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: Intel: Skylake: add HDA BE DAIs" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: Intel: Skylake: add HDA BE DAIs has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3d17871349d5cec0a37ce9407ba72fdbf8572cfd Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Date: Wed, 22 Aug 2018 15:25:01 -0500 Subject: [PATCH] ASoC: Intel: Skylake: add HDA BE DAIs Add support for HDA BE DAIs in SKL platform driver. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-pcm.c | 70 +++++++++++++++++++++++++------ 1 file changed, 58 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e39103edd..00b7a91b18c9 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -32,6 +32,7 @@ #define HDA_MONO 1 #define HDA_STEREO 2 #define HDA_QUAD 4 +#define HDA_MAX 8 static const struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | @@ -569,7 +570,10 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, stream_tag = hdac_stream(link_dev)->stream_tag; /* set the stream tag in the codec dai dma params */ - snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); + else + snd_soc_dai_set_tdm_slot(codec_dai, 0, stream_tag, 0, 0); p_params.s_fmt = snd_pcm_format_width(params_format(params)); p_params.ch = params_channels(params); @@ -995,21 +999,63 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { - .name = "HD-Codec Pin", + .name = "Analog CPU DAI", .ops = &skl_link_dai_ops, .playback = { - .stream_name = "HD-Codec Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { - .stream_name = "HD-Codec Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .stream_name = "Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Alt Analog CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Alt Analog CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Alt Analog CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, +}, +{ + .name = "Digital CPU DAI", + .ops = &skl_link_dai_ops, + .playback = { + .stream_name = "Digital CPU Playback", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Digital CPU Capture", + .channels_min = HDA_MONO, + .channels_max = HDA_MAX, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, }, }, }; -- 2.18.0

1 0

[alsa-devel] [PATCH v8 0/7] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 28 Aug '18

28 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v8 is based on Marks' for-next branch - the merge of topic/drm_audio_component is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: 1. fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) 2. when the screen goes blank, HDMI playback stops. This isn't related to this patchset since we rely on the existing hdac_hdmi codec, but will need additional investigations. Changes v8: (comments from Takashi) - move hda_codec.h to include/sound (from sound/pci/hda) - remove unnecessary includes (local.h) - remove Kconfig option to control HDaudio suppport Changes v7 - fix route count for headless boards such as Up2 to enable all 3 iDisp routes (only iDisp3 was set-up due to a bad macro). Changes v6 (comments from Mark and Takashi) - simplify Kconfig options, remove FORCE for now and make code simpler - change SPDIX style to C++ for C files - add log message if invalid configuration reached - remove static int counter, added private counter in context instead - fixed a couple of indendation issues to make checkpatch.pl --strict happy Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ALSA: hda: move hda_codec.h to include/sound ASoC: Intel: common: add table for HDA-based platforms Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers {sound/pci/hda => include/sound}/hda_codec.h | 0 include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/hda_bind.c | 14 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_jack.c | 2 +- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_ca0110.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_cmedia.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- sound/pci/hda/patch_via.c | 2 +- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 484 ++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 182 +++++++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 ++- sound/soc/intel/skylake/skl.c | 96 +++- sound/soc/intel/skylake/skl.h | 12 +- 40 files changed, 1109 insertions(+), 51 deletions(-) rename {sound/pci/hda => include/sound}/hda_codec.h (100%) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.17.1

2 9

[alsa-devel] [PATCH v2] ASoC: sigmadsp: safeload should not have lower byte limit
by Danny Smith 28 Aug '18

28 Aug '18

Fixed range in safeload conditional to allow safeload to up to 20 bytes, without a lower limit. Signed-off-by: Danny Smith <dannys(a)axis.com> --- sound/soc/codecs/sigmadsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index d53680ac78e4..6df158669420 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -117,8 +117,7 @@ static int sigmadsp_ctrl_write(struct sigmadsp *sigmadsp, struct sigmadsp_control *ctrl, void *data) { /* safeload loads up to 20 bytes in a atomic operation */ - if (ctrl->num_bytes > 4 && ctrl->num_bytes <= 20 && sigmadsp->ops && - sigmadsp->ops->safeload) + if (ctrl->num_bytes <= 20 && sigmadsp->ops && sigmadsp->ops->safeload) return sigmadsp->ops->safeload(sigmadsp, ctrl->addr, data, ctrl->num_bytes); else -- 2.11.0

3 2

[alsa-devel] Applied "ASoC: adau17x1: Implemented safeload support" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: adau17x1: Implemented safeload support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b0f2d651299f0743818bdadcbe6a67d7869e0da1 Mon Sep 17 00:00:00 2001 From: Danny Smith <dannys(a)axis.com> Date: Tue, 21 Aug 2018 13:07:49 +0200 Subject: [PATCH] ASoC: adau17x1: Implemented safeload support Safeload support has been implemented which is used when updating for instance filter parameters using alsa controls. Without safeload support audio can become distorted during update. Signed-off-by: Danny Smith <dannys(a)axis.com> Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 77 ++++++++++++++++++++++++++++++++++++- 1 file changed, 75 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..c0bc429249fa 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -21,11 +21,18 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> #include <linux/regmap.h> +#include <asm/unaligned.h> #include "sigmadsp.h" #include "adau17x1.h" #include "adau-utils.h" +#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006 +#define ADAU17X1_SAFELOAD_TRIGGER 0x0007 +#define ADAU17X1_SAFELOAD_DATA 0x0001 +#define ADAU17X1_SAFELOAD_DATA_SIZE 20 +#define ADAU17X1_WORD_SIZE 4 + static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", }; @@ -326,6 +333,17 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static bool adau17x1_has_safeload(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1781: + return true; + default: + return false; + } +} + static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { @@ -957,6 +975,56 @@ int adau17x1_resume(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(adau17x1_resume); +static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + uint8_t buf[ADAU17X1_WORD_SIZE]; + uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE]; + unsigned int addr_offset; + unsigned int nbr_words; + int ret; + + /* write data to safeload addresses. Check if len is not a multiple of + * 4 bytes, if so we need to zero pad. + */ + nbr_words = len / ADAU17X1_WORD_SIZE; + if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) { + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, bytes, len); + } else { + nbr_words++; + memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE); + memcpy(data, bytes, len); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, data, + nbr_words * ADAU17X1_WORD_SIZE); + } + + if (ret < 0) + return ret; + + /* Write target address, target address is offset by 1 */ + addr_offset = addr - 1; + put_unaligned_be32(addr_offset, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + /* write nbr of words to trigger address */ + put_unaligned_be32(nbr_words, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + return 0; +} + +static const struct sigmadsp_ops adau17x1_sigmadsp_ops = { + .safeload = adau17x1_safeload, +}; + int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name) @@ -1002,8 +1070,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); if (firmware_name) { - adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, - firmware_name); + if (adau17x1_has_safeload(adau)) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + &adau17x1_sigmadsp_ops, firmware_name); + } else { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + NULL, firmware_name); + } if (IS_ERR(adau->sigmadsp)) { dev_warn(dev, "Could not find firmware file: %ld\n", PTR_ERR(adau->sigmadsp)); -- 2.18.0

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: rt5670: Add quirk for Thinkpad 8 tablet
by Hans de Goede 28 Aug '18

28 Aug '18

The Thinkpad 8 needs a quirk for jack-detect and the internal mic to work correctly. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 732ef928b25d..455fe7cff700 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2875,6 +2875,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_DEV_GPIO | RT5670_JD_MODE1), }, + { + .callback = rt5670_quirk_cb, + .ident = "Lenovo Thinkpad Tablet 8", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad 8"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_DEV_GPIO | + RT5670_JD_MODE1), + }, { .callback = rt5670_quirk_cb, .ident = "Lenovo Thinkpad Tablet 10", -- 2.18.0

2 5

[alsa-devel] [PATCH] ASoC: rsnd: Add r8a774a1 support
by Fabrizio Castro 28 Aug '18

28 Aug '18

Document RZ/G2M (R8A774A1) SoC bindings. Signed-off-by: Fabrizio Castro <fabrizio.castro(a)bp.renesas.com> Reviewed-by: Biju Das <biju.das(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 9e764270..8083f0d 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -340,10 +340,11 @@ Required properties: - compatible : "renesas,rcar_sound-<soctype>", fallbacks "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 + "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) Examples with soctypes are: - "renesas,rcar_sound-r8a7743" (RZ/G1M) - "renesas,rcar_sound-r8a7745" (RZ/G1E) + - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7779" (R-Car H1) - "renesas,rcar_sound-r8a7790" (R-Car H2) -- 2.7.4

4 3

[alsa-devel] [PATCH] ASoC: AMD: Set constraints for DMIC and MAX98357a codec
by Akshu Agrawal 28 Aug '18

28 Aug '18

We support dual channel, 48Khz. This constraint was set only for da7219. It is being extended to DMIC and MAX98357a. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 066d5489..cf2f648 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -162,10 +162,21 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) static int cz_max_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + /* + * 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); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -177,20 +188,42 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) static int cz_dmic0_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + /* + * 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); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } static int cz_dmic1_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + /* + * 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); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: AMD: Change MCLK to 48Mhz
by Akshu Agrawal 28 Aug '18

28 Aug '18

25Mhz MCLK which was earlier used was of spread type. Thus, we were not getting accurate rate. The 48Mhz system clk is of non-spread type and we are changing to it to get accurate rate. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index cf2f648..55d7f61 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -42,7 +42,7 @@ #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" -#define CZ_PLAT_CLK 25000000 +#define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: atmel: tse850: switch to SPDX license identifier
by Peter Rosin 28 Aug '18

28 Aug '18

Convert to // comments in the leading comment, drop the boilerplate license text and use the correct MODULE_LICENSE. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/tse850-pcm5142.c | 78 +++++++++++++++++++--------------------- 1 file changed, 36 insertions(+), 42 deletions(-) diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 3a1393283156..214adcad5419 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -1,44 +1,38 @@ -/* - * TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec - * - * Copyright (C) 2016 Axentia Technologies AB - * - * Author: Peter Rosin <peda(a)axentia.se> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -/* - * loop1 relays - * IN1 +---o +------------+ o---+ OUT1 - * \ / - * + + - * | / | - * +--o +--. | - * | add | | - * | V | - * | .---. | - * DAC +----------->|Sum|---+ - * | '---' | - * | | - * + + - * - * IN2 +---o--+------------+--o---+ OUT2 - * loop2 relays - * - * The 'loop1' gpio pin controlls two relays, which are either in loop - * position, meaning that input and output are directly connected, or - * they are in mixer position, meaning that the signal is passed through - * the 'Sum' mixer. Similarly for 'loop2'. - * - * In the above, the 'loop1' relays are inactive, thus feeding IN1 to the - * mixer (if 'add' is active) and feeding the mixer output to OUT1. The - * 'loop2' relays are active, short-cutting the TSE-850 from channel 2. - * IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name - * of the (filtered) output from the PCM5142 codec. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// TSE-850 audio - ASoC driver for the Axentia TSE-850 with a PCM5142 codec +// +// Copyright (C) 2016 Axentia Technologies AB +// +// Author: Peter Rosin <peda(a)axentia.se> +// +// loop1 relays +// IN1 +---o +------------+ o---+ OUT1 +// \ / +// + + +// | / | +// +--o +--. | +// | add | | +// | V | +// | .---. | +// DAC +----------->|Sum|---+ +// | '---' | +// | | +// + + +// +// IN2 +---o--+------------+--o---+ OUT2 +// loop2 relays +// +// The 'loop1' gpio pin controlls two relays, which are either in loop +// position, meaning that input and output are directly connected, or +// they are in mixer position, meaning that the signal is passed through +// the 'Sum' mixer. Similarly for 'loop2'. +// +// In the above, the 'loop1' relays are inactive, thus feeding IN1 to the +// mixer (if 'add' is active) and feeding the mixer output to OUT1. The +// 'loop2' relays are active, short-cutting the TSE-850 from channel 2. +// IN1, IN2, OUT1 and OUT2 are TSE-850 connectors and DAC is the PCB name +// of the (filtered) output from the PCM5142 codec. #include <linux/clk.h> #include <linux/gpio.h> @@ -452,4 +446,4 @@ module_platform_driver(tse850_driver); /* Module information */ MODULE_AUTHOR("Peter Rosin <peda(a)axentia.se>"); MODULE_DESCRIPTION("ALSA SoC driver for TSE-850 with PCM5142 codec"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- 2.11.0

2 1

[alsa-devel] Suspected heap overflow in snd_midi_event_encode_byte()
by Prashant Malani 28 Aug '18

28 Aug '18

Hi, The Chromium fuzzers detected a potential heap overflow in snd_midi_event_encode_byte() when attempting to encode an invalid data sequence. The potential bug was observed in alsa-lib-1.1.5 (the source seems similar to alsa-lib-1.1.6 so it is likely present there too). Code to reproduce (condensed to fit in an email): std::array<int, 4> arr{ {0x0A, 0x0B, 0x0C, 0x0D} }; snd_midi_event_t* encoder; snd_midi_event_new(arr.size(), &encoder); for (int i = 0; i < arr.size(); i++) { snd_seq_event_t event; int result = snd_midi_event_encode_byte(encoder, arr[i], &event); if (result < 0) { // Log error and return.... } if ( result == 1) { // Send the completed message and return. } } .... The first call to snd_midi_event_encode_byte() using byte 0x0A causes the |encoder->qlen| to underflow and become a large unsigned value, and |encoder->read| to become 2. Subsequent bytes processed will get written to the |encoder->buf| buffer, with |dev->read| getting incremented after every byte. As a result, by the time we get to byte 0x0D, |encoder->read| is already 4, and this results in a heap overflow (relevant line in alsa-lib is src/seq/seq_midi_event.c:425) I'm not sure if the above input array is a valid input to snd_midi_event_encode_byte(), but I'm guessing we should be doing some error checking to make sure we're not processing incorrect/unexpected/invalid bytes. Any suggestions about how one can submit a fix for this? Best regards, Prashant

3 7

[alsa-devel] [PATCH v2] ASoC: Convert to using %pOFn instead of device_node.name
by Rob Herring 28 Aug '18

28 Aug '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Carlo Caione <carlo(a)caione.org> Cc: Kevin Hilman <khilman(a)baylibre.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-amlogic(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - stm32_sai_sub: Unwrap now shorter line sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 +++---- 5 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c1d1d06783e5..57b484768a58 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); + snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b52223..9981668ab590 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, of_node_put(dma_channel_np); return ret; } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn", + (unsigned long long) res.start, dma_channel_np); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2914ba0d965b..b76a5f4f1785 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -478,7 +478,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, ret = axg_card_set_link_name(card, link, "be"); if (ret) - dev_err(card->dev, "error setting %s link name\n", np->name); + dev_err(card->dev, "error setting %pOFn link name\n", np); return ret; } diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index f22654253c43..d597eba61992 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!pdev) { dev_err(&sai_client->pdev->dev, - "Device not found for node %s\n", np_provider->name); + "Device not found for node %pOFn\n", np_provider); return -ENODEV; } diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 06fba9650ac4..56a227e0bd71 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1124,16 +1124,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sync = SAI_SYNC_NONE; if (args.np) { if (args.np == np) { - dev_err(&pdev->dev, "%s sync own reference\n", - np->name); + dev_err(&pdev->dev, "%pOFn sync own reference\n", np); of_node_put(args.np); return -EINVAL; } sai->np_sync_provider = of_get_parent(args.np); if (!sai->np_sync_provider) { - dev_err(&pdev->dev, "%s parent node not found\n", - np->name); + dev_err(&pdev->dev, "%pOFn parent node not found\n", + np); of_node_put(args.np); return -ENODEV; } -- 2.17.1

2 1

[alsa-devel] [PATCH] [sound] hdac-codec runtime suspended at PM:Suspend.
by Anshuman Gupta 28 Aug '18

28 Aug '18

Keep hdac-codec to be in runtime suspended while entering to suspend. If hdac-codec is already in runtime suspend state skip its power down sequence in prepare, power up sequence in complete phase. Avoid resuming hdac controller PCI device 00:1f.3 from runtime suspend state in case hdac-codec already in runtime-suspend state, this is unnecessary and block the direct complete even for hdac controller PCI device 00:1f.3. This enabled direct complete path for hdac-codec and PCI device 00:1f.3. Signed-off-by: Anshuman Gupta <anshuman.gupta(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index f3b4f4d..810a8a6 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1852,6 +1852,8 @@ static int hdmi_codec_prepare(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; + if (pm_runtime_status_suspended(dev)) + return 1; pm_runtime_get_sync(&edev->hdac.dev); /* @@ -1873,6 +1875,8 @@ static void hdmi_codec_complete(struct device *dev) struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_device *hdac = &edev->hdac; + if (pm_runtime_status_suspended(dev)) + return; /* Power up afg */ snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); -- 2.7.4

3 6

[alsa-devel] [PATCH] ALSA: seq: Do error checks at creating system ports
by Takashi Iwai 28 Aug '18

28 Aug '18

snd_seq_system_client_init() doesn't check the errors returned from its port creations. Let's do it properly and handle the error paths. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_system.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index 3b89b0adad6a..0dc5d5a45ecc 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void) { struct snd_seq_port_callback pcallbacks; struct snd_seq_port_info *port; + int err; port = kzalloc(sizeof(*port), GFP_KERNEL); if (!port) @@ -148,7 +149,10 @@ int __init snd_seq_system_client_init(void) port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; port->addr.client = sysclient; port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER; - snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port); + err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, + port); + if (err < 0) + goto error_port; /* register announcement port */ strcpy(port->name, "Announce"); @@ -158,16 +162,24 @@ int __init snd_seq_system_client_init(void) port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; port->addr.client = sysclient; port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; - snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port); + err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, + port); + if (err < 0) + goto error_port; announce_port = port->addr.port; kfree(port); return 0; + + error_port: + snd_seq_system_client_done(); + kfree(port); + return err; } /* unregister our internal client */ -void __exit snd_seq_system_client_done(void) +void snd_seq_system_client_done(void) { int oldsysclient = sysclient; -- 2.18.0

1 0

[alsa-devel] [PATCH] [ALSA] seq: add error check in snd_seq_system_client_init()
by Dan Carpenter 28 Aug '18

28 Aug '18

Static checkers complain that snd_seq_create_kernel_client() can return -EBUSY here so we need to have some error handling. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index 8ce1d0b40dce..3b89b0adad6a 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -134,6 +134,10 @@ int __init snd_seq_system_client_init(void) /* register client */ sysclient = snd_seq_create_kernel_client(NULL, 0, "System"); + if (sysclient < 0) { + kfree(port); + return sysclient; + } /* register timer */ strcpy(port->name, "Timer");

2 1

[alsa-devel] i.Mx35 SGTL5000 - audio hangs only on system reboot.
by Nikhil 28 Aug '18

28 Aug '18

Hello, I am facing an issue with SGTL5000 codec which happens only on reboot. Below are the details. On doing a power cycle audio works fine, but if I do a "reboot", I could see that driver probe is success, but on playing a file using ogg, it hangs. At the moment, I am not sure where the issue is. I did some search in Freescale/NXP forums and found there are issues related to SGTL5000 and i2c bus on reboot, but in my case, on reboot the probe itself is success, so I believe , its not the I2c reboot issue. Any help as to how I can debug further or solutions is much appreciated. Thanks in advance. Regards Nikhil System details: chip : i.Mx35 Kernel : 4.14 linux-fslc DeviceTree : sound { compatible = "fsl,imx-audio-sgtl5000"; model = "ebox-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT", "Ext Spk", "LINE_OUT"; mux-int-port = <1>; mux-ext-port = <4>; }; codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sgtl5000>; reg = <0x0a>; clocks = <&clks 86>; VDDA-supply = <&vcam_reg>; VDDIO-supply = <&reg_sgtl5000_vddd>; }; sgtl output: $ dmesg | grep sgtl [ 2.416167] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 2.425204] sgtl5000 1-000a: Using internal LDO instead of VDDD: check ER1 [ 2.562911] imx-sgtl5000 sound: sgtl5000 <-> 43fa0000.ssi mapping ok [ 3.247742] #0: ebox-sgtl5000 dma output on reboot: $ dmesg | grep -i dma [ 0.049859] DMA: preallocated 256 KiB pool for atomic coherent allocations [ 0.710313] i2c i2c-1: can't use DMA, using PIO instead. [ 0.712794] i2c i2c-3: can't use DMA, using PIO instead. [ 0.818858] ipu-core ipu-core: private_candidate: dma0chan0 filter said false [ 0.818918] ipu-core ipu-core: private_candidate: dma0chan1 filter said false [ 0.818945] ipu-core ipu-core: private_candidate: dma0chan2 filter said false [ 0.818972] ipu-core ipu-core: private_candidate: dma0chan3 filter said false [ 0.818997] ipu-core ipu-core: private_candidate: dma0chan4 filter said false [ 0.819022] ipu-core ipu-core: private_candidate: dma0chan5 filter said false [ 0.819047] ipu-core ipu-core: private_candidate: dma0chan6 filter said false [ 0.819072] ipu-core ipu-core: private_candidate: dma0chan7 filter said false [ 0.819098] ipu-core ipu-core: private_candidate: dma0chan8 filter said false [ 0.819123] ipu-core ipu-core: private_candidate: dma0chan9 filter said false [ 0.819148] ipu-core ipu-core: private_candidate: dma0chan10 filter said false [ 0.819174] ipu-core ipu-core: private_candidate: dma0chan11 filter said false [ 0.819199] ipu-core ipu-core: private_candidate: dma0chan12 filter said false [ 0.819224] ipu-core ipu-core: private_candidate: dma0chan13 filter said false [ 0.961113] ipu-core ipu-core: IDMAC_CONF 0x70, IC_CONF 0x0, IDMAC_CHA_EN 0x0, IDMAC_CHA_PRI 0x0, IDMAC_CHA_BUSY 0x0 [ 0.961409] dma dma0chan14: Found channel 0xe, irq 176 [ 0.961442] dmaengine: __dma_request_channel: success (dma0chan14) [ 0.988705] dma dma0chan14: Submitting sg d79f3b94 [ 0.988776] dma dma0chan14: Updated sg d79f3b94 on channel 0xe buffer 0 [ 0.988827] ipu-core ipu-core: IDMAC_CONF 0x70, IC_CONF 0x0, IDMAC_CHA_EN 0x0, IDMAC_CHA_PRI 0x4000, IDMAC_CHA_BUSY 0x0 [ 0.988885] ipu-core ipu-core: IDMAC_CONF 0x70, IC_CONF 0x0, IDMAC_CHA_EN 0x4000, IDMAC_CHA_PRI 0x4000, IDMAC_CHA_BUSY 0x0 [ 1.084868] imx-sdma 53fd4000.sdma: no event needs to be remapped [ 1.085941] imx-sdma 53fd4000.sdma: loaded firmware 1.0 [ 1.934535] spi_imx 43fa4000.cspi: dma setup error -19, use pio [ 2.401179] mmc0: SDHCI controller on 53fb4000.esdhc [53fb4000.esdhc] using DMA [ 2.472063] dmaengine: __dma_request_channel: success (dma1chan0) [ 2.472236] imx-sdma 53fd4000.sdma: private_candidate: dma1chan0 busy [ 2.472374] dmaengine: __dma_request_channel: success (dma1chan1) sgtl5000 registers on reboot , but before playing sound file: $ cat /sys/kernel/debug/regmap/1-000a/registers 000: a011 002: 0060 004: 0008 006: 0080 00a: 0010 00e: 020c 010: 3c3c 014: 015f 020: 0000 022: 1818 024: 0022 026: 0008 028: 01c1 02a: 0200 02c: 031c 02e: 0f0f 030: 7a60 032: 5000 034: 0800 036: 0017 03a: 0000 03c: 0000 100: 0000 102: 0000 104: 8040 106: 051f 108: 0000 10a: 0040 10c: 0000 10e: 0000 110: 0000 116: 002f 118: 002f 11a: 002f 11c: 002f 11e: 002f 120: 8000 122: 0000 124: 5100 126: 1473 128: 0028 12a: 0050 12c: 0000 12e: 0000 130: 0000 132: 0000 134: 0000 136: 0000 138: 0000 13a: 0000 ssi registers on reboot , but before playing sound file: $ cat /sys/kernel/debug/regmap/43fa0000.ssi/registers 00: 00000000 04: 00000000 10: 00001058 18: 00003003 1c: 0000020d 20: 0000020d 24: 00040100 28: 00040100 2c: 00880088 30: 00001111 34: 00000000 38: 00000000 48: 00000000 4c: 00000000 50: 00000000 54: 00000000 58: 00000000 sgtl5000 registers on reboot , after playing sound file: $ cat /sys/kernel/debug/regmap/1-000a/registers 000: a011 002: 0061 004: 0007 006: 0090 00a: 0010 00e: 0200 010: 3c3c 014: 015f 020: 0000 022: 1818 024: 0022 026: 0008 028: 01c1 02a: 0200 02c: 031c 02e: 0f0f 030: 7ff9 032: 786c 034: 0808 036: 0017 03a: 0000 03c: 0000 100: 0000 102: 0000 104: 8040 106: 051f 108: 0000 10a: 0040 10c: 0000 10e: 0000 110: 0000 116: 002f 118: 002f 11a: 002f 11c: 002f 11e: 002f 120: 8000 122: 0000 124: 5100 126: 1473 128: 0028 12a: 0050 12c: 0000 12e: 0000 130: 0000 132: 0000 134: 0000 136: 0000 138: 0000 13a: 0000 ssi registers on reboot , after playing sound file: $ cat /sys/kernel/debug/regmap/43fa0000.ssi/registers 00: 00ffff00 04: 00000000 10: 0000105b 18: 01d035f7 1c: 0000028d 20: 0000028d 24: 00056100 28: 00040100 2c: 00880888 30: 00001111 34: 00000000 38: 00000000 48: 00000000 4c: 00000000 50: 00000000 54: 00000000 58: 00000000 dma output after playing a sound file: [ 115.626358] imx-sdma 53fd4000.sdma: load_address = 834 [ 115.626482] imx-sdma 53fd4000.sdma: wml = 0x00000020 [ 115.626512] imx-sdma 53fd4000.sdma: shp_addr = 0x43fa0000 [ 115.626537] imx-sdma 53fd4000.sdma: per_addr = 0x00000000 [ 115.626563] imx-sdma 53fd4000.sdma: event_mask0 = 0x20000000 [ 115.626588] imx-sdma 53fd4000.sdma: event_mask1 = 0x00000000 [ 116.072831] imx-sdma 53fd4000.sdma: sdma_prep_dma_cyclic channel: 1 [ 116.072956] imx-sdma 53fd4000.sdma: load_address = 834 [ 116.072984] imx-sdma 53fd4000.sdma: wml = 0x00000020 [ 116.073009] imx-sdma 53fd4000.sdma: shp_addr = 0x43fa0000 [ 116.073032] imx-sdma 53fd4000.sdma: per_addr = 0x00000000 [ 116.073055] imx-sdma 53fd4000.sdma: event_mask0 = 0x20000000 [ 116.073077] imx-sdma 53fd4000.sdma: event_mask1 = 0x00000000 [ 116.073160] imx-sdma 53fd4000.sdma: entry 0: count: 1768 dma: 0x97b80000 intr [ 116.073191] imx-sdma 53fd4000.sdma: entry 1: count: 1768 dma: 0x97b806e8 intr [ 116.073219] imx-sdma 53fd4000.sdma: entry 2: count: 1768 dma: 0x97b80dd0 intr [ 116.073247] imx-sdma 53fd4000.sdma: entry 3: count: 1768 dma: 0x97b814b8 wrap int

1 0

[alsa-devel] [PATCH] ASoC: Convert to using %pOFn instead of device_node.name
by Rob Herring 28 Aug '18

28 Aug '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Carlo Caione <carlo(a)caione.org> Cc: Kevin Hilman <khilman(a)baylibre.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Cc: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)st.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-amlogic(a)lists.infradead.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 8 ++++---- 5 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c1d1d06783e5..57b484768a58 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); + snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b52223..9981668ab590 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, of_node_put(dma_channel_np); return ret; } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn", + (unsigned long long) res.start, dma_channel_np); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2914ba0d965b..b76a5f4f1785 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -478,7 +478,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, ret = axg_card_set_link_name(card, link, "be"); if (ret) - dev_err(card->dev, "error setting %s link name\n", np->name); + dev_err(card->dev, "error setting %pOFn link name\n", np); return ret; } diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index f22654253c43..d597eba61992 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!pdev) { dev_err(&sai_client->pdev->dev, - "Device not found for node %s\n", np_provider->name); + "Device not found for node %pOFn\n", np_provider); return -ENODEV; } diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 06fba9650ac4..cad8c7888292 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1124,16 +1124,16 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sync = SAI_SYNC_NONE; if (args.np) { if (args.np == np) { - dev_err(&pdev->dev, "%s sync own reference\n", - np->name); + dev_err(&pdev->dev, "%pOFn sync own reference\n", + np); of_node_put(args.np); return -EINVAL; } sai->np_sync_provider = of_get_parent(args.np); if (!sai->np_sync_provider) { - dev_err(&pdev->dev, "%s parent node not found\n", - np->name); + dev_err(&pdev->dev, "%pOFn parent node not found\n", + np); of_node_put(args.np); return -ENODEV; } -- 2.17.1

2 1

[alsa-devel] [PATCH] sound: Convert to using %pOFn instead of device_node.name
by Rob Herring 28 Aug '18

28 Aug '18

In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/aoa/soundbus/i2sbus/core.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 000b58522106..bf32e1c74748 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -157,18 +157,19 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct device_node *child = NULL, *sound = NULL; struct resource *r; int i, layout = 0, rlen, ok = force; - static const char *rnames[] = { "i2sbus: %s (control)", - "i2sbus: %s (tx)", - "i2sbus: %s (rx)" }; + char node_name[6]; + static const char *rnames[] = { "i2sbus: %pOFn (control)", + "i2sbus: %pOFn (tx)", + "i2sbus: %pOFn (rx)" }; static irq_handler_t ints[] = { i2sbus_bus_intr, i2sbus_tx_intr, i2sbus_rx_intr }; - if (strlen(np->name) != 5) + if (snprintf(node_name, sizeof(node_name), "%pOFn", np) != 5) return 0; - if (strncmp(np->name, "i2s-", 4)) + if (strncmp(np, "i2s-", 4)) return 0; dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL); @@ -228,13 +229,13 @@ static int i2sbus_add_dev(struct macio_dev *macio, dev->sound.pcmid = -1; dev->macio = macio; dev->control = control; - dev->bus_number = np->name[4] - 'a'; + dev->bus_number = node_name[4] - 'a'; INIT_LIST_HEAD(&dev->sound.codec_list); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { dev->interrupts[i] = -1; snprintf(dev->rnames[i], sizeof(dev->rnames[i]), - rnames[i], np->name); + rnames[i], np); } for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { int irq = irq_of_parse_and_map(np, i); -- 2.17.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: cs4265: SOC_SINGLE register value error fix
by Matt Flax 28 Aug '18

28 Aug '18

From: Matt Flax <flatmax(a)flatmax.org> The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..15b4ae04 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.17.1

2 3

Re: [alsa-devel] INFO: rcu detected stall in snd_seq_write
by Dmitry Vyukov 28 Aug '18

28 Aug '18

On Mon, Aug 27, 2018 at 10:10 AM, syzbot <syzbot+97aae04ce27e39cbfca9(a)syzkaller.appspotmail.com> wrote: > Hello, > > syzbot found the following crash on: > > HEAD commit: e27bc174c9c6 Add linux-next specific files for 20180824 > git tree: linux-next > console output: https://syzkaller.appspot.com/x/log.txt?x=16e0823e400000 > kernel config: https://syzkaller.appspot.com/x/.config?x=28446088176757ea > dashboard link: https://syzkaller.appspot.com/bug?extid=97aae04ce27e39cbfca9 > compiler: gcc (GCC) 8.0.1 20180413 (experimental) > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1756c50e400000 This seems to have been attributed to some x86 file, I will take a look as to why. +ALSA maintainers > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+97aae04ce27e39cbfca9(a)syzkaller.appspotmail.com > > 8021q: adding VLAN 0 to HW filter on device team0 > 8021q: adding VLAN 0 to HW filter on device team0 > 8021q: adding VLAN 0 to HW filter on device team0 > 8021q: adding VLAN 0 to HW filter on device team0 > 8021q: adding VLAN 0 to HW filter on device team0 > rcu: INFO: rcu_sched self-detected stall on CPU > rcu: 0-....: (105000 ticks this GP) idle=2f6/1/0x4000000000000002 > softirq=23001/23001 fqs=26239 > rcu: (t=105008 jiffies g=49145 q=2382) > NMI backtrace for cpu 0 > CPU: 0 PID: 8551 Comm: syz-executor7 Not tainted 4.18.0-next-20180824+ #47 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS > Google 01/01/2011 > Call Trace: > <IRQ> > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0x1c9/0x2b4 lib/dump_stack.c:113 > nmi_cpu_backtrace.cold.3+0x48/0x88 lib/nmi_backtrace.c:101 > nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 > arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 > trigger_single_cpu_backtrace include/linux/nmi.h:162 [inline] > rcu_dump_cpu_stacks+0x175/0x1c2 kernel/rcu/tree.c:1340 > print_cpu_stall.cold.78+0x2fb/0x59c kernel/rcu/tree.c:1478 > check_cpu_stall kernel/rcu/tree.c:1550 [inline] > __rcu_pending kernel/rcu/tree.c:3276 [inline] > rcu_pending kernel/rcu/tree.c:3319 [inline] > rcu_check_callbacks+0xd4a/0x15a0 kernel/rcu/tree.c:2665 > update_process_times+0x2d/0x70 kernel/time/timer.c:1636 > tick_sched_handle+0x9f/0x180 kernel/time/tick-sched.c:164 > tick_sched_timer+0x45/0x130 kernel/time/tick-sched.c:1274 > __run_hrtimer kernel/time/hrtimer.c:1398 [inline] > __hrtimer_run_queues+0x3eb/0xff0 kernel/time/hrtimer.c:1460 > hrtimer_interrupt+0x2f3/0x750 kernel/time/hrtimer.c:1518 > local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1029 [inline] > smp_apic_timer_interrupt+0x16d/0x6a0 arch/x86/kernel/apic/apic.c:1054 > apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:867 > </IRQ> > RIP: 0010:lock_release+0xa2/0x9f0 kernel/locking/lockdep.c:3910 > Code: f2 c7 40 08 f2 f2 f2 f2 c7 40 0c 00 f2 f2 f2 c7 40 10 f2 f2 f2 f2 c7 > 40 14 00 f2 f2 f2 65 48 8b 04 25 28 00 00 00 48 89 45 d0 <31> c0 48 89 f8 48 > c1 e8 03 0f b6 14 10 48 89 f8 83 e0 07 83 c0 03 > RSP: 0018:ffff8801aa0078c8 EFLAGS: 00000292 ORIG_RAX: ffffffffffffff13 > RAX: 2ec5cd2493d34200 RBX: 1ffff10035400f1e RCX: 0000000000000000 > RDX: dffffc0000000000 RSI: 0000000000000000 RDI: ffff8801a9b56bbc > RBP: ffff8801aa0079f8 R08: 00000000000010df R09: 0000000000000001 > R10: ffff8801a9b56be8 R11: 737c3e5c87308161 R12: ffff8801aa0079d0 > R13: ffff8801c778d4c8 R14: ffff8801aa007ae0 R15: ffff8801a9b56380 > __might_fault+0x19e/0x1e0 mm/memory.c:4585 > _copy_from_user+0x30/0x150 lib/usercopy.c:10 > copy_from_user include/linux/uaccess.h:147 [inline] > snd_seq_write+0x472/0x8d0 sound/core/seq/seq_clientmgr.c:1033 > __vfs_write+0x117/0x9d0 fs/read_write.c:485 > vfs_write+0x1fc/0x560 fs/read_write.c:549 > ksys_write+0x101/0x260 fs/read_write.c:598 > __do_sys_write fs/read_write.c:610 [inline] > __se_sys_write fs/read_write.c:607 [inline] > __x64_sys_write+0x73/0xb0 fs/read_write.c:607 > do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 > entry_SYSCALL_64_after_hwframe+0x49/0xbe > RIP: 0033:0x457089 > Code: fd b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 > 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff > 0f 83 cb b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 > RSP: 002b:00007f09af65dc78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 > RAX: ffffffffffffffda RBX: 00007f09af65e6d4 RCX: 0000000000457089 > RDX: 00000000ffffff76 RSI: 0000000020000000 RDI: 0000000000000003 > RBP: 00000000009300a0 R08: 0000000000000000 R09: 0000000000000000 > R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff > R13: 00000000004d78a8 R14: 00000000004ca886 R15: 0000000000000000 > > > --- > This bug is generated by a bot. It may contain errors. > See https://goo.gl/tpsmEJ for more information about syzbot. > syzbot engineers can be reached at syzkaller(a)googlegroups.com. > > syzbot will keep track of this bug report. See: > https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with > syzbot. > syzbot can test patches for this bug, for details see: > https://goo.gl/tpsmEJ#testing-patches > > -- > You received this message because you are subscribed to the Google Groups > "syzkaller-bugs" group. > To unsubscribe from this group and stop receiving emails from it, send an > email to syzkaller-bugs+unsubscribe(a)googlegroups.com. > To view this discussion on the web visit > https://groups.google.com/d/msgid/syzkaller-bugs/000000000000e5050205746dcb…. > For more options, visit https://groups.google.com/d/optout.

1 0

[alsa-devel] INFO: task hung in snd_seq_ioctl
by syzbot 28 Aug '18

28 Aug '18

Hello, syzbot found the following crash on: HEAD commit: e27bc174c9c6 Add linux-next specific files for 20180824 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=11bf156a400000 kernel config: https://syzkaller.appspot.com/x/.config?x=28446088176757ea dashboard link: https://syzkaller.appspot.com/bug?extid=b059d98be99f7b009662 compiler: gcc (GCC) 8.0.1 20180413 (experimental) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+b059d98be99f7b009662(a)syzkaller.appspotmail.com overlayfs: unrecognized mount option "/dev/snapshot" or missing value overlayfs: unrecognized mount option "/dev/snapshot" or missing value INFO: task syz-executor0:9532 blocked for more than 140 seconds. Not tainted 4.18.0-next-20180824+ #47 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor0 D25944 9532 9598 0x00000004 Call Trace: context_switch kernel/sched/core.c:2825 [inline] __schedule+0x87c/0x1df0 kernel/sched/core.c:3473 schedule+0xfb/0x450 kernel/sched/core.c:3517 schedule_preempt_disabled+0x10/0x20 kernel/sched/core.c:3575 __mutex_lock_common kernel/locking/mutex.c:1003 [inline] __mutex_lock+0xbf9/0x1700 kernel/locking/mutex.c:1073 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1088 snd_seq_ioctl+0x221/0x440 sound/core/seq/seq_clientmgr.c:2137 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:501 [inline] do_vfs_ioctl+0x1de/0x1720 fs/ioctl.c:685 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:702 __do_sys_ioctl fs/ioctl.c:709 [inline] __se_sys_ioctl fs/ioctl.c:707 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:707 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: fd b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f01a1a09c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 00007f01a1a0a6d4 RCX: 0000000000457089 RDX: 00000000200000c0 RSI: 00000000c08c5332 RDI: 0000000000000004 RBP: 0000000000930320 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004cffa0 R14: 00000000004bf048 R15: 0000000000000004 Showing all locks held in the system: 1 lock held by khungtaskd/775: #0: 00000000e99a49ba (rcu_read_lock){....}, at: debug_show_all_locks+0xd0/0x428 kernel/locking/lockdep.c:4436 1 lock held by rsyslogd/4337: 2 locks held by getty/4428: #0: 0000000057084618 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 000000009ad3c5cd (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4429: #0: 00000000a879cc50 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 00000000cec89b30 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4430: #0: 000000002f98cec0 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 00000000a64abc28 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4431: #0: 000000001aa7e298 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 00000000fc7196ae (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4432: #0: 000000001245086c (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 0000000053d9216f (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4433: #0: 000000009dbd25fa (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 000000002c7d4de5 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4434: #0: 00000000b1fc4e5b (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 00000000410ff32a (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by kworker/0:2/4760: 1 lock held by syz-executor0/9471: 1 lock held by syz-executor0/9532: #0: 000000001c41a5c8 (&client->ioctl_mutex){+.+.}, at: snd_seq_ioctl+0x221/0x440 sound/core/seq/seq_clientmgr.c:2137 ============================================= NMI backtrace for cpu 0 CPU: 0 PID: 775 Comm: khungtaskd Not tainted 4.18.0-next-20180824+ #47 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c9/0x2b4 lib/dump_stack.c:113 nmi_cpu_backtrace.cold.3+0x48/0x88 lib/nmi_backtrace.c:101 nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_all_cpu_backtrace include/linux/nmi.h:144 [inline] check_hung_uninterruptible_tasks kernel/hung_task.c:204 [inline] watchdog+0xb39/0x1040 kernel/hung_task.c:265 kthread+0x35a/0x420 kernel/kthread.c:246 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:415 Sending NMI from CPU 0 to CPUs 1: INFO: NMI handler (nmi_cpu_backtrace_handler) took too long to run: 1.031 msecs NMI backtrace for cpu 1 CPU: 1 PID: 9471 Comm: syz-executor0 Not tainted 4.18.0-next-20180824+ #47 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:trace_lock_acquire include/trace/events/lock.h:13 [inline] RIP: 0010:lock_acquire+0x3cc/0x4f0 kernel/locking/lockdep.c:3900 Code: ff 0d 38 3e a2 7e e9 e0 fd ff ff 65 ff 05 2c 3e a2 7e 48 ba 00 00 00 00 00 fc ff df 48 8d 45 98 48 c1 e8 03 48 01 d0 c6 00 00 <48> 8b 15 e5 5c 27 07 48 89 55 98 c6 00 f8 e8 21 b8 06 00 85 c0 74 RSP: 0018:ffff8801db107a38 EFLAGS: 00000082 RAX: ffffed003b620f58 RBX: 1ffff1003b620f4c RCX: 0000000000000000 RDX: dffffc0000000000 RSI: 0000000000000000 RDI: ffff88018d882afc RBP: ffff8801db107b28 R08: 0000000000000001 R09: 0000000000000000 R10: ffffed003b6246de R11: ffff8801db1236f3 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000001 FS: 00007f01a1a8e700(0000) GS:ffff8801db100000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffffffff600400 CR3: 00000001b29e9000 CR4: 00000000001406e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <IRQ> __raw_spin_lock_irq include/linux/spinlock_api_smp.h:128 [inline] _raw_spin_lock_irq+0x5e/0x80 kernel/locking/spinlock.c:160 __run_hrtimer kernel/time/hrtimer.c:1400 [inline] __hrtimer_run_queues+0x443/0xff0 kernel/time/hrtimer.c:1460 hrtimer_interrupt+0x2f3/0x750 kernel/time/hrtimer.c:1518 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1029 [inline] smp_apic_timer_interrupt+0x16d/0x6a0 arch/x86/kernel/apic/apic.c:1054 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:867 </IRQ> RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:783 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0xa1/0xc0 kernel/locking/spinlock.c:184 Code: 68 bc f1 87 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 80 3c 02 00 75 21 48 83 3d 2e be 3a 01 00 74 0e 48 89 df 57 9d <0f> 1f 44 00 00 eb bb 0f 0b 0f 0b e8 cf 10 05 fb eb 97 e8 c8 10 05 RSP: 0018:ffff88013d4c77b0 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: dffffc0000000000 RBX: 0000000000000282 RCX: 1ffff10031b10564 RDX: 1ffffffff0fe378d RSI: 0000000000000000 RDI: 0000000000000282 RBP: ffff88013d4c77c0 R08: ffff88018d882b00 R09: 0000000000000006 R10: ffff88018d8822c0 R11: 0000000000000000 R12: ffffffff8861fc20 R13: ffff8801b3d27780 R14: 0000000000000282 R15: ffffffff8a34a020 spin_unlock_irqrestore include/linux/spinlock.h:384 [inline] snd_seq_client_use_ptr+0x9e/0x3f0 sound/core/seq/seq_clientmgr.c:178 snd_seq_dispatch_event+0xbf/0x650 sound/core/seq/seq_clientmgr.c:845 snd_seq_check_queue.part.4+0x139/0x360 sound/core/seq/seq_queue.c:275 snd_seq_check_queue sound/core/seq/seq_queue.c:255 [inline] snd_seq_enqueue_event+0x346/0x4d0 sound/core/seq/seq_queue.c:343 snd_seq_client_enqueue_event+0x2a5/0x510 sound/core/seq/seq_clientmgr.c:957 snd_seq_write+0x3f1/0x8d0 sound/core/seq/seq_clientmgr.c:1074 __vfs_write+0x117/0x9d0 fs/read_write.c:485 vfs_write+0x1fc/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x73/0xb0 fs/read_write.c:607 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: fd b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f01a1a8dc78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f01a1a8e6d4 RCX: 0000000000457089 RDX: 00000000ffffff76 RSI: 0000000020000000 RDI: 0000000000000003 RBP: 00000000009300a0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004d78a8 R14: 00000000004ca886 R15: 0000000000000000 --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

1 0

[alsa-devel] INFO: task hung in snd_seq_write
by syzbot 28 Aug '18

28 Aug '18

Hello, syzbot found the following crash on: HEAD commit: e27bc174c9c6 Add linux-next specific files for 20180824 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=1772020a400000 kernel config: https://syzkaller.appspot.com/x/.config?x=28446088176757ea dashboard link: https://syzkaller.appspot.com/bug?extid=4c595632b98bb8ffcc66 compiler: gcc (GCC) 8.0.1 20180413 (experimental) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+4c595632b98bb8ffcc66(a)syzkaller.appspotmail.com INFO: task syz-executor1:8015 blocked for more than 140 seconds. Not tainted 4.18.0-next-20180824+ #47 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor1 D25408 8015 4471 0x00000004 Call Trace: context_switch kernel/sched/core.c:2825 [inline] __schedule+0x87c/0x1df0 kernel/sched/core.c:3473 schedule+0xfb/0x450 kernel/sched/core.c:3517 schedule_preempt_disabled+0x10/0x20 kernel/sched/core.c:3575 __mutex_lock_common kernel/locking/mutex.c:1003 [inline] __mutex_lock+0xbf9/0x1700 kernel/locking/mutex.c:1073 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1088 snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 __vfs_write+0x117/0x9d0 fs/read_write.c:485 vfs_write+0x1fc/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x73/0xb0 fs/read_write.c:607 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: 08 e8 3b a9 fb ff 48 8b 44 24 20 48 8b 4c 24 18 48 8b 54 24 38 48 8b 7c 24 48 eb 8d 48 8b 6c 24 50 48 83 c4 58 c3 48 89 04 24 <48> 89 5c 24 08 e8 0d a9 fb ff 48 8b 44 24 40 48 8b 4c 24 48 48 8b RSP: 002b:00007f5f1ec8fc78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f5f1ec906d4 RCX: 0000000000457089 RDX: 0000000000000020 RSI: 0000000020000480 RDI: 0000000000000006 RBP: 0000000000930140 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004d7388 R14: 00000000004c1780 R15: 0000000000000001 INFO: task syz-executor1:8019 blocked for more than 140 seconds. Not tainted 4.18.0-next-20180824+ #47 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor1 D25896 8019 4471 0x00000004 Call Trace: context_switch kernel/sched/core.c:2825 [inline] __schedule+0x87c/0x1df0 kernel/sched/core.c:3473 schedule+0xfb/0x450 kernel/sched/core.c:3517 schedule_preempt_disabled+0x10/0x20 kernel/sched/core.c:3575 __mutex_lock_common kernel/locking/mutex.c:1003 [inline] __mutex_lock+0xbf9/0x1700 kernel/locking/mutex.c:1073 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1088 snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 __vfs_write+0x117/0x9d0 fs/read_write.c:485 vfs_write+0x1fc/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x73/0xb0 fs/read_write.c:607 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: 08 e8 3b a9 fb ff 48 8b 44 24 20 48 8b 4c 24 18 48 8b 54 24 38 48 8b 7c 24 48 eb 8d 48 8b 6c 24 50 48 83 c4 58 c3 48 89 04 24 <48> 89 5c 24 08 e8 0d a9 fb ff 48 8b 44 24 40 48 8b 4c 24 48 48 8b RSP: 002b:00007f5f1ec6ec78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f5f1ec6f6d4 RCX: 0000000000457089 RDX: 0000000000000090 RSI: 0000000020000580 RDI: 0000000000000003 RBP: 00000000009301e0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004d7340 R14: 00000000004ca576 R15: 0000000000000002 INFO: task syz-executor1:8026 blocked for more than 140 seconds. Not tainted 4.18.0-next-20180824+ #47 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor1 D25112 8026 4471 0x00000004 Call Trace: context_switch kernel/sched/core.c:2825 [inline] __schedule+0x87c/0x1df0 kernel/sched/core.c:3473 schedule+0xfb/0x450 kernel/sched/core.c:3517 schedule_preempt_disabled+0x10/0x20 kernel/sched/core.c:3575 __mutex_lock_common kernel/locking/mutex.c:1003 [inline] __mutex_lock+0xbf9/0x1700 kernel/locking/mutex.c:1073 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1088 snd_seq_ioctl+0x221/0x440 sound/core/seq/seq_clientmgr.c:2137 vfs_ioctl fs/ioctl.c:46 [inline] file_ioctl fs/ioctl.c:501 [inline] do_vfs_ioctl+0x1de/0x1720 fs/ioctl.c:685 ksys_ioctl+0xa9/0xd0 fs/ioctl.c:702 __do_sys_ioctl fs/ioctl.c:709 [inline] __se_sys_ioctl fs/ioctl.c:707 [inline] __x64_sys_ioctl+0x73/0xb0 fs/ioctl.c:707 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: 08 e8 3b a9 fb ff 48 8b 44 24 20 48 8b 4c 24 18 48 8b 54 24 38 48 8b 7c 24 48 eb 8d 48 8b 6c 24 50 48 83 c4 58 c3 48 89 04 24 <48> 89 5c 24 08 e8 0d a9 fb ff 48 8b 44 24 40 48 8b 4c 24 48 48 8b RSP: 002b:00007f5f1ec4dc78 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 00007f5f1ec4e6d4 RCX: 0000000000457089 RDX: 00000000200006c0 RSI: 00000000c08c5332 RDI: 0000000000000006 RBP: 0000000000930280 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004cfed0 R14: 00000000004bf048 R15: 0000000000000003 INFO: task syz-executor1:8040 blocked for more than 140 seconds. Not tainted 4.18.0-next-20180824+ #47 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor1 D25896 8040 4471 0x00000004 Call Trace: context_switch kernel/sched/core.c:2825 [inline] __schedule+0x87c/0x1df0 kernel/sched/core.c:3473 schedule+0xfb/0x450 kernel/sched/core.c:3517 schedule_preempt_disabled+0x10/0x20 kernel/sched/core.c:3575 __mutex_lock_common kernel/locking/mutex.c:1003 [inline] __mutex_lock+0xbf9/0x1700 kernel/locking/mutex.c:1073 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1088 snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 __vfs_write+0x117/0x9d0 fs/read_write.c:485 vfs_write+0x1fc/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x73/0xb0 fs/read_write.c:607 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: 08 e8 3b a9 fb ff 48 8b 44 24 20 48 8b 4c 24 18 48 8b 54 24 38 48 8b 7c 24 48 eb 8d 48 8b 6c 24 50 48 83 c4 58 c3 48 89 04 24 <48> 89 5c 24 08 e8 0d a9 fb ff 48 8b 44 24 40 48 8b 4c 24 48 48 8b RSP: 002b:00007f5f1ec2cc78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f5f1ec2d6d4 RCX: 0000000000457089 RDX: 00000000000000a0 RSI: 00000000200004c0 RDI: 0000000000000006 RBP: 0000000000930320 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004d7418 R14: 00000000004c1870 R15: 0000000000000004 INFO: task syz-executor1:8043 blocked for more than 140 seconds. Not tainted 4.18.0-next-20180824+ #47 "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. syz-executor1 D25896 8043 4471 0x00000004 Call Trace: context_switch kernel/sched/core.c:2825 [inline] __schedule+0x87c/0x1df0 kernel/sched/core.c:3473 schedule+0xfb/0x450 kernel/sched/core.c:3517 schedule_preempt_disabled+0x10/0x20 kernel/sched/core.c:3575 __mutex_lock_common kernel/locking/mutex.c:1003 [inline] __mutex_lock+0xbf9/0x1700 kernel/locking/mutex.c:1073 mutex_lock_nested+0x16/0x20 kernel/locking/mutex.c:1088 snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 __vfs_write+0x117/0x9d0 fs/read_write.c:485 vfs_write+0x1fc/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x73/0xb0 fs/read_write.c:607 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: 08 e8 3b a9 fb ff 48 8b 44 24 20 48 8b 4c 24 18 48 8b 54 24 38 48 8b 7c 24 48 eb 8d 48 8b 6c 24 50 48 83 c4 58 c3 48 89 04 24 <48> 89 5c 24 08 e8 0d a9 fb ff 48 8b 44 24 40 48 8b 4c 24 48 48 8b RSP: 002b:00007f5f1ebeac78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f5f1ebeb6d4 RCX: 0000000000457089 RDX: 00000000ffffff76 RSI: 0000000020000000 RDI: 0000000000000006 RBP: 0000000000930460 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004d7790 R14: 00000000004ca7f1 R15: 0000000000000006 Showing all locks held in the system: 1 lock held by khungtaskd/776: #0: 0000000098b2492f (rcu_read_lock){....}, at: debug_show_all_locks+0xd0/0x428 kernel/locking/lockdep.c:4436 1 lock held by rsyslogd/4329: 2 locks held by getty/4419: #0: 00000000a4856276 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 000000004b3fbd4e (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4420: #0: 00000000d3776216 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 0000000018b91ba2 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4421: #0: 00000000e90a323e (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 00000000369e6155 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4422: #0: 000000001139f51f (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 000000003aa121ca (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4423: #0: 00000000dc25f399 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 000000009e2aa14c (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4424: #0: 00000000fe22db15 (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 00000000249dec83 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 2 locks held by getty/4425: #0: 000000002fbf78cf (&tty->ldisc_sem){++++}, at: ldsem_down_read+0x37/0x40 drivers/tty/tty_ldsem.c:353 #1: 000000000377af90 (&ldata->atomic_read_lock){+.+.}, at: n_tty_read+0x335/0x1ce0 drivers/tty/n_tty.c:2140 1 lock held by syz-executor1/7988: 1 lock held by syz-executor1/8015: #0: 000000007dc4af24 (&client->ioctl_mutex){+.+.}, at: snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 1 lock held by syz-executor1/8019: #0: 000000007dc4af24 (&client->ioctl_mutex){+.+.}, at: snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 1 lock held by syz-executor1/8026: #0: 000000007dc4af24 (&client->ioctl_mutex){+.+.}, at: snd_seq_ioctl+0x221/0x440 sound/core/seq/seq_clientmgr.c:2137 1 lock held by syz-executor1/8040: #0: 000000007dc4af24 (&client->ioctl_mutex){+.+.}, at: snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 1 lock held by syz-executor1/8043: #0: 000000007dc4af24 (&client->ioctl_mutex){+.+.}, at: snd_seq_write+0x1f1/0x8d0 sound/core/seq/seq_clientmgr.c:1021 ============================================= NMI backtrace for cpu 1 CPU: 1 PID: 776 Comm: khungtaskd Not tainted 4.18.0-next-20180824+ #47 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c9/0x2b4 lib/dump_stack.c:113 nmi_cpu_backtrace.cold.3+0x48/0x88 lib/nmi_backtrace.c:101 nmi_trigger_cpumask_backtrace+0x151/0x192 lib/nmi_backtrace.c:62 arch_trigger_cpumask_backtrace+0x14/0x20 arch/x86/kernel/apic/hw_nmi.c:38 trigger_all_cpu_backtrace include/linux/nmi.h:144 [inline] check_hung_uninterruptible_tasks kernel/hung_task.c:204 [inline] watchdog+0xb39/0x1040 kernel/hung_task.c:265 kthread+0x35a/0x420 kernel/kthread.c:246 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:415 Sending NMI from CPU 1 to CPUs 0: INFO: NMI handler (nmi_cpu_backtrace_handler) took too long to run: 1.319 msecs NMI backtrace for cpu 0 CPU: 0 PID: 7988 Comm: syz-executor1 Not tainted 4.18.0-next-20180824+ #47 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:mark_lock+0x181/0x19f0 kernel/locking/lockdep.c:3155 Code: 43 10 00 00 00 00 48 8b 75 d0 65 48 33 34 25 28 00 00 00 0f 85 f5 11 00 00 48 81 c4 08 01 00 00 5b 41 5c 41 5d 41 5e 41 5f 5d <c3> 48 c7 c7 a0 97 90 89 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 RSP: 0018:ffff8801db0072e8 EFLAGS: 00000086 RAX: 0000000000000001 RBX: 0000000000000039 RCX: 1ffff100365035f1 RDX: 1ffffffff12853f6 RSI: 0000000000000000 RDI: ffffffff89429fb0 RBP: ffff8801db007680 R08: 0000000000000008 R09: 0000000000000001 R10: ffff8801b281a700 R11: ffff8801b281af2c R12: ffff8801b281af8a R13: 0000000000000040 R14: ffff8801b281a700 R15: 000000000000000e FS: 00007f5f1ecb1700(0000) GS:ffff8801db000000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: ffffffffff600400 CR3: 00000001c3883000 CR4: 00000000001406f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <IRQ> lock_acquire+0x1e4/0x4f0 kernel/locking/lockdep.c:3901 seqcount_lockdep_reader_access include/linux/seqlock.h:81 [inline] read_seqcount_begin include/linux/seqlock.h:164 [inline] ktime_get+0x14e/0x440 kernel/time/timekeeping.c:754 hrtimer_forward_now include/linux/hrtimer.h:473 [inline] perf_swevent_hrtimer+0x405/0x510 kernel/events/core.c:9124 __run_hrtimer kernel/time/hrtimer.c:1398 [inline] __hrtimer_run_queues+0x3eb/0xff0 kernel/time/hrtimer.c:1460 hrtimer_interrupt+0x2f3/0x750 kernel/time/hrtimer.c:1518 local_apic_timer_interrupt arch/x86/kernel/apic/apic.c:1029 [inline] smp_apic_timer_interrupt+0x16d/0x6a0 arch/x86/kernel/apic/apic.c:1054 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:867 </IRQ> RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:783 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0xa1/0xc0 kernel/locking/spinlock.c:184 Code: 68 bc f1 87 48 b8 00 00 00 00 00 fc ff df 48 89 fa 48 c1 ea 03 80 3c 02 00 75 21 48 83 3d 2e be 3a 01 00 74 0e 48 89 df 57 9d <0f> 1f 44 00 00 eb bb 0f 0b 0f 0b e8 cf 10 05 fb eb 97 e8 c8 10 05 RSP: 0018:ffff88019430f7b0 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: dffffc0000000000 RBX: 0000000000000282 RCX: 1ffff100365035ec RDX: 1ffffffff0fe378d RSI: 0000000000000000 RDI: 0000000000000282 RBP: ffff88019430f7c0 R08: ffff8801b281af40 R09: 0000000000000006 R10: ffff8801b281a700 R11: 0000000000000000 R12: ffffffff8861fc20 R13: ffff8801b0156540 R14: 0000000000000282 R15: ffffffff8a34a020 spin_unlock_irqrestore include/linux/spinlock.h:384 [inline] snd_seq_client_use_ptr+0x9e/0x3f0 sound/core/seq/seq_clientmgr.c:178 snd_seq_dispatch_event+0xbf/0x650 sound/core/seq/seq_clientmgr.c:845 snd_seq_check_queue.part.4+0x139/0x360 sound/core/seq/seq_queue.c:275 snd_seq_check_queue sound/core/seq/seq_queue.c:255 [inline] snd_seq_enqueue_event+0x346/0x4d0 sound/core/seq/seq_queue.c:343 snd_seq_client_enqueue_event+0x2a5/0x510 sound/core/seq/seq_clientmgr.c:957 snd_seq_write+0x3f1/0x8d0 sound/core/seq/seq_clientmgr.c:1074 __vfs_write+0x117/0x9d0 fs/read_write.c:485 vfs_write+0x1fc/0x560 fs/read_write.c:549 ksys_write+0x101/0x260 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x73/0xb0 fs/read_write.c:607 do_syscall_64+0x1b9/0x820 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x457089 Code: fd b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 cb b4 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f5f1ecb0c78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 00007f5f1ecb16d4 RCX: 0000000000457089 RDX: 00000000ffffff76 RSI: 0000000020000000 RDI: 0000000000000003 RBP: 00000000009300a0 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000ffffffff R13: 00000000004d7790 R14: 00000000004ca7f1 R15: 0000000000000000 --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot.

1 0

[alsa-devel] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write
by George Anchev 27 Aug '18

27 Aug '18

Hi, Each time I boot in journalctl I notice the lines which you can see in the attached journal.txt file. Attaching also the output of alsa-info.sh. Both files are compressed to minimize message size. Please let me know if this is an actual bug and what is causing it. P. S. The "Bug Tracker" link on project's home page shows a "not found" page. -- George

2 4

[alsa-devel] snd_usb_usx2y holding onto a lock?
by Joseph Spencer 27 Aug '18

27 Aug '18

When plugging an audio interface in with nomodeset I see the following in the logs: Aug 08 23:19:23 localhost.localdomain kernel: INFO: task pulseaudio:1750 blocked for more than 120 seconds. Aug 08 23:19:23 localhost.localdomain kernel: Tainted: G OE 4.17.11-200.fc28.x86_64+debug #1 Aug 08 23:19:23 localhost.localdomain kernel: "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. Aug 08 23:19:23 localhost.localdomain kernel: pulseaudio D12408 1750 1268 0x00000000 Aug 08 23:19:23 localhost.localdomain kernel: Call Trace: Aug 08 23:19:23 localhost.localdomain kernel: ? __schedule+0x2ba/0xaf0 Aug 08 23:19:23 localhost.localdomain kernel: ? prepare_to_wait_event+0xd2/0x180 Aug 08 23:19:23 localhost.localdomain kernel: schedule+0x2f/0x90 Aug 08 23:19:23 localhost.localdomain kernel: usX2Y_urbs_start+0x208/0x470 [snd_usb_usx2y] Aug 08 23:19:23 localhost.localdomain kernel: ? finish_wait+0x80/0x80 Aug 08 23:19:23 localhost.localdomain kernel: snd_usX2Y_pcm_prepare+0xb4/0xd0 [snd_usb_usx2y] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_do_prepare+0x19/0x30 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_action_single+0x36/0x80 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_action_nonatomic+0x61/0x70 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_common_ioctl+0x3d2/0xb70 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: ? selinux_file_ioctl+0x171/0x210 Aug 08 23:19:23 localhost.localdomain kernel: snd_pcm_ioctl+0x23/0x30 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: do_vfs_ioctl+0xa5/0x6d0 Aug 08 23:19:23 localhost.localdomain kernel: ksys_ioctl+0x60/0x90 Aug 08 23:19:23 localhost.localdomain kernel: __x64_sys_ioctl+0x16/0x20 Aug 08 23:19:23 localhost.localdomain kernel: do_syscall_64+0x60/0x1f0 Aug 08 23:19:23 localhost.localdomain kernel: entry_SYSCALL_64_after_hwframe+0x49/0xbe Aug 08 23:19:23 localhost.localdomain kernel: RIP: 0033:0x7f59021c4ec7 Aug 08 23:19:23 localhost.localdomain kernel: RSP: 002b:00007ffd83d81448 EFLAGS: 00000206 ORIG_RAX: 0000000000000010 Aug 08 23:19:23 localhost.localdomain kernel: RAX: ffffffffffffffda RBX: 0000556d69312220 RCX: 00007f59021c4ec7 Aug 08 23:19:23 localhost.localdomain kernel: RDX: 0000556d6931ae20 RSI: 0000000000004140 RDI: 0000000000000016 Aug 08 23:19:23 localhost.localdomain kernel: RBP: 00007ffd83d81b60 R08: 0000000000000000 R09: 0000000000001138 Aug 08 23:19:23 localhost.localdomain kernel: R10: 0000000000000004 R11: 0000000000000206 R12: 00007ffd83d81830 Aug 08 23:19:23 localhost.localdomain kernel: R13: 00007ffd83d81cf4 R14: 0000000000000000 R15: 00007ffd83d815c0 Aug 08 23:19:23 localhost.localdomain kernel: Showing all locks held in the system: Aug 08 23:19:23 localhost.localdomain kernel: 1 lock held by khungtaskd/38: Aug 08 23:19:23 localhost.localdomain kernel: #0: 00000000b04056c7 (tasklist_lock){.+.+}, at: debug_show_all_locks+0x3e/0x165 Aug 08 23:19:23 localhost.localdomain kernel: 2 locks held by pulseaudio/1750: Aug 08 23:19:23 localhost.localdomain kernel: #0: 000000008d30dafb (snd_pcm_link_rwsem){++++}, at: snd_pcm_action_nonatomic+0x1e/0x70 [snd_pcm] Aug 08 23:19:23 localhost.localdomain kernel: #1: 00000000c8fc4ab6 (&usX2Y(card)->pcm_mutex){+.+.}, at: snd_usX2Y_pcm_prepare+0x38/0xd0 [snd_usb_usx2y] The audio interface doesn't work and without nomodeset X freezes. Any help pointing me in the right direction would be greatly appreciated. It looks like an issue with the snd_usb_usx2y driver to me, but I'm not sure. Joe Spencer

2 3

[alsa-devel] [PATCH] ALSA: pcm: signedness bug in snd_pcm_plug_alloc()
by Dan Carpenter 27 Aug '18

27 Aug '18

The "frames" variable is unsigned so the error handling doesn't work properly. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 0391cb1a4f19..141c5f3a9575 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames);

2 1

[alsa-devel] [PATCH] ALSA: hdspm: fix spelling mistake "Initializeing" -> "Initializing"
by Colin King 27 Aug '18

27 Aug '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_dbg message and also remove extraneous white space and repeated question marks. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 11b5b5e0e058..679ad0415e3b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6534,7 +6534,7 @@ static int snd_hdspm_create_alsa_devices(struct snd_card *card, dev_dbg(card->dev, "Update mixer controls...\n"); hdspm_update_simple_mixer_controls(hdspm); - dev_dbg(card->dev, "Initializeing complete ???\n"); + dev_dbg(card->dev, "Initializing complete?\n"); err = snd_card_register(card); if (err < 0) { -- 2.17.1

2 1

[alsa-devel] [PATCH 0/8] ALSA: non-cached memalloc helper
by Takashi Iwai 27 Aug '18

27 Aug '18

Hi, this is a patchset to add the non-cached page handling in the memory allocation helpers, which have been implemented individually in several driver codes. This also fixes a long-time bug of HD-audio non-mmap behavior in the non-snoop mode, which was reported recently. Takashi === Takashi Iwai (8): ALSA: memalloc: Don't align the size to power-of-two ALSA: memalloc: Simplify snd_malloc_dev_pages() calls ALSA: memalloc: Add non-cached buffer type ALSA: hda: Check the non-cached stream buffers more explicitly ALSA: hda: Use new non-cached allocation for non-snoop mode ALSA: hda: Remove substream allocation/free ops ALSA: intel8x0: Use the new non-cached allocation for 440MX workaround ALSA: intel_hdmi: Use the new non-cached allocation include/sound/memalloc.h | 3 + sound/core/memalloc.c | 40 ++++++++----- sound/core/sgbuf.c | 15 ++++- sound/pci/hda/hda_controller.c | 13 +++-- sound/pci/hda/hda_controller.h | 7 +-- sound/pci/hda/hda_intel.c | 103 ++++----------------------------- sound/pci/hda/hda_tegra.c | 18 +----- sound/pci/intel8x0.c | 89 ++++------------------------ sound/x86/intel_hdmi_audio.c | 29 +--------- 9 files changed, 75 insertions(+), 242 deletions(-) -- 2.18.0

1 8

[alsa-devel] [PATCH v3] ALSA: pcm: Return 0 when size < start_threshold in capture
by Ricardo Biehl Pasquali 26 Aug '18

26 Aug '18

In __snd_pcm_lib_xfer(), when capture, if state is PREPARED and size is less than start_threshold nothing can be done. As there is no error, 0 is returned. Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- version 3: Set 'err' variable inside 'else' block. --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2173,11 +2173,16 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(str goto _end_unlock; if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED && - size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) + runtime->status->state == SNDRV_PCM_STATE_PREPARED) { + if (size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) + goto _end_unlock; + } else { + /* nothing to do */ + err = 0; goto _end_unlock; + } } runtime->twake = runtime->control->avail_min ? : 1;

2 1

[alsa-devel] [PATCH 2/2] ASoC: rt5682: Change DAC/ADC volume scale
by shumingf＠realtek.com 24 Aug '18

24 Aug '18

From: Shuming Fan <shumingf(a)realtek.com> The step of DAC/ADC volume scale changes from 0.375dB to 0.75dB Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 0ea2759..731c6a8 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -750,8 +750,8 @@ static bool rt5682_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ @@ -1114,7 +1114,7 @@ static const struct snd_kcontrol_new rt5682_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682_DAC1_DIG_VOL, - RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 86, 0, dac_vol_tlv), /* IN Boost Volume */ SOC_SINGLE_TLV("CBJ Boost Volume", RT5682_CBJ_BST_CTRL, @@ -1124,7 +1124,7 @@ static const struct snd_kcontrol_new rt5682_snd_controls[] = { SOC_DOUBLE("STO1 ADC Capture Switch", RT5682_STO1_ADC_DIG_VOL, RT5682_L_MUTE_SFT, RT5682_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5682_STO1_ADC_DIG_VOL, - RT5682_L_VOL_SFT, RT5682_R_VOL_SFT, 127, 0, adc_vol_tlv), + RT5682_L_VOL_SFT + 1, RT5682_R_VOL_SFT + 1, 63, 0, adc_vol_tlv), /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682_STO1_ADC_BOOST, -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: max98373: Added 10ms delay after amp software reset
by Ryan Lee 24 Aug '18

24 Aug '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Applied 10ms delay after amp software reset. 10ms guard time is required for stability. sound/soc/codecs/max98373.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 2764fae..6aaff03 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -730,6 +730,7 @@ static int max98373_probe(struct snd_soc_component *component) /* Software Reset */ regmap_write(max98373->regmap, MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); + mdelay(10); /* IV default slot configuration */ regmap_write(max98373->regmap, @@ -818,6 +819,7 @@ static int max98373_resume(struct device *dev) regmap_write(max98373->regmap, MAX98373_R2000_SW_RESET, MAX98373_SOFT_RESET); + mdelay(10); regcache_cache_only(max98373->regmap, false); regcache_sync(max98373->regmap); return 0; -- 2.7.4

3 4

Re: [alsa-devel] [RFC] hda/realtek filtered headphone output
by Takashi Iwai 23 Aug '18

23 Aug '18

[ corrected to the right ML ] On Thu, 23 Aug 2018 14:37:39 +0200, René Rebe wrote: > > Hi, > > On 23 Aug 2018, at 14:31, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Thu, 23 Aug 2018 13:10:39 +0200, > > René Rebe wrote: > >> > >> Hi Takashi, > >> > >> I uploaded the requested log but did not hear back. > >> The limited headphone frequency response (aka high-pass) is still an issue with 4.18. > >> I rebased the patch: > >> > >> https://svn.exactcode.de/t2/trunk/package/base/linux/hda-realtek.patch > >> > >> This affects more devices (e.g. Dell XPS 13, ThinkPads) any maybe many more which I do not have at hand. > >> > >> Further guidance how to proceed with this issue to get it fixed upstream is welcome. > > > > As mentioned, a better way is to define the preferred mapping table. > > Take a look at alc274_fixup_bind_dacs() as an example > > Did you had a chance to look at the log I uploaded to get an idea if it is possible to improve this auto parser to figure out the connection? Did you take a look at the function? It's pretty straightforward, you can specify the mapping you'd need there. Takashi > > René > > > Takashi > > > >> > >> René > >> > >> On 06 Mar 2018, at 16:07, René Rebe <rene(a)exactcode.de> wrote: > >> > >>> Hi Takashi, > >>> > >>> On 03/05/2018 12:55 PM, Takashi Iwai wrote: > >>>> On Sat, 03 Mar 2018 13:06:49 +0100, > >>>> René Rebe wrote: > >>>>> > >>>>> Hi all, > >>>>> > >>>>> testing some new laptops, I noticed the headphone frequency response was lacking > >>>>> bass. First I noticed this in the Lenovo T470s, and later I found this also be the case > >>>>> on Surface Pro 2 and 3. The Dell XPS13 may also be affected, but I had to return it > >>>>> before I found out the solution on the other machines: > >>>>> > >>>>> https://www.youtube.com/watch?v=a1YkPtfC4LI > >>>>> > >>>>> I had not yet sent the patch because a) I was too busy and b) I wanted to review this further. > >>>>> > >>>>> As the other day I got a Lenovo ideapad 720s to test with exactly the same issue, here is the > >>>>> current hacked patch: > >>>>> > >>>>> https://svn.exactcode.de/t2/trunk/package/base/linux/surface-audio.patch > >>>>> > >>>>> However, given that all newer realtek codecs that came on my desk are affected, > >>>>> maybe there is something wrong with the hda_auto_parser.c - I did not yet fully grasp > >>>>> all the logic of that. > >>>>> > >>>>> As you can see in the patch and video (…) the problem is the signal routing of NID 0x15 > >>>>> or 0x21 (depending whether it is an older, or newer codec). This is the headphone out, > >>>>> and as far as I can see the auto_parser assigned the index 0 pcm output (NID 0x0c) > >>>>> which result in this filtered frequency response, likely intended for the laptop’s speakers? > >>>>> Using the next NID (I guess that was 0xd?) results in a full frequency response and at > >>>>> least to my ears is immediately hearable and obvious. > >>>>> > >>>>> Without this fixed routing the headphone sounds like a PC speaker in a coke can. > >>>>> > >>>>> So the question for the authors and HDA experts: Can this be fixed in the auto_parser, > >>>>> or is my patch (which btw. has to small unrelated cleanups) modulo potentially some > >>>>> renaming of the enum and function, now that more than the Surface is affected, the > >>>>> way to go? If this can be fixed more generally I would be happy to hear! > >>>> Could you give the output of alsa-info.sh? We need to check the > >>>> details of the codec topology. > >>> > >>> To avoid sending the huge output to the list, I pressed the upload button, and it gave me this Url: > >>> > >>> > >>> http://www.alsa-project.org/db/?f=c68cad9c06aa999f017f1e43b52448d778d2903f > >>> > >>> This is with the linked patch, if you want to see it without the change, I need to re-build my kernel again :-/ > >>> > >>>> In general, the right approach is to fix at parsing the routing, not > >>>> fiddling with the codec routing dynamically at initialization. > >>>> Currently the easy fix would be to provide the fixed DAC connection > >>>> list. Maybe we can enhance the generic parser to receive some hints > >>>> for DAC selection. The routes are determined by calculating "badness" > >>>> values per route, so in theory, we can give some badness values > >>>> depending on the target pin type (headphone, speaker, etc). > >>> > >>> Let me know if I should try some alternative solution, ... > >>> > >>> René > >>> > >>> -- > >>> René Rebe, ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > >>> https://exactcode.com | https://t2sde.org | https://rene.rebe.de > >> > >> -- > >> ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > >> DE Legal: Amtsgericht Berlin (Charlottenburg) HRB 105123B, Tax-ID#: DE251602478 > >> Managing Director: René Rebe > >> http://exactcode.com | http://exactscan.com | http://ocrkit.com | http://t2-project.org | http://rene.rebe.de > >> > > -- > ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin > DE Legal: Amtsgericht Berlin (Charlottenburg) HRB 105123B, Tax-ID#: DE251602478 > Managing Director: René Rebe > http://exactcode.com | http://exactscan.com | http://ocrkit.com | http://t2-project.org | http://rene.rebe.de >

2 2

[alsa-devel] [PATCH v7 0/7] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 22 Aug '18

22 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v7 is based on Marks' for-next branch - the merge of topic/drm_audio_component is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: 1. fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) 2. when the screen goes blank, HDMI playback stops. This isn't related to this patchset since we rely on the existing hdac_hdmi codec, but will need additional investigations. Changes v7 - fix route count for headless boards such as Up2 to enable all 3 iDisp routes (only iDisp3 was set-up due to a bad macro). Changes v6 (comments from Mark and Takashi) - simplify Kconfig options, remove FORCE for now and make code simpler - change SPDIX style to C++ for C files - add log message if invalid configuration reached - remove static int counter, added private counter in context instead - fixed a couple of indendation issues to make checkpatch.pl --strict happy Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 485 ++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/intel/Kconfig | 9 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 182 +++++++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 ++- sound/soc/intel/skylake/skl.c | 104 +++- sound/soc/intel/skylake/skl.h | 12 +- 17 files changed, 1103 insertions(+), 27 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.17.1

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[alsa-devel] [PATCH 01/11] ASoC: SOF: Add Sound Open Firmware driver core
by Liam Girdwood 22 Aug '18

22 Aug '18

The Sound Open Firmware driver core is a generic architecture independent layer that allows SOF to be used on many different different architectures and platforms. It abstracts DSP operations and IO methods so that the target DSP can be an internal memory mapped or external SPI or I2C based device. This abstraction also allows SOF to be run on many different VMs on the same physical HW. SOF also requires some data in ASoC PCM runtime data for looking up SOF data during ASoC PCM operations. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc.h | 1 + include/sound/sof.h | 92 +++++++ sound/soc/sof/core.c | 373 +++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 504 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 970 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/sof-priv.h diff --git a/include/sound/soc.h b/include/sound/soc.h index a4915148f739..df8e9084876b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1133,6 +1133,7 @@ struct snd_soc_pcm_runtime { /* runtime devices */ struct snd_pcm *pcm; struct snd_compr *compr; + struct snd_sof_pcm *sof; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; diff --git a/include/sound/sof.h b/include/sound/sof.h new file mode 100644 index 000000000000..835d0ae4770c --- /dev/null +++ b/include/sound/sof.h @@ -0,0 +1,92 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2017 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> + */ + +#ifndef __INCLUDE_SOUND_SOF_H +#define __INCLUDE_SOUND_SOF_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/pci.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <uapi/sound/sof-ipc.h> + +struct snd_sof_dsp_ops; + +/* SOF probe type */ +enum sof_device_type { + SOF_DEVICE_PCI = 0, + SOF_DEVICE_APCI, + SOF_DEVICE_SPI +}; + +/* + * SOF Platform data. + */ +struct snd_sof_pdata { + u32 id; /* PCI/ACPI ID */ + const struct firmware *fw; + const char *drv_name; + const char *name; + + /* parent device */ + struct device *dev; + enum sof_device_type type; + + /* descriptor */ + const struct sof_dev_desc *desc; + + /* machine */ + struct platform_device *pdev_mach; + const struct snd_soc_acpi_mach *machine; +}; + +/* + * Descriptor used for setting up SOF platform data. This is used when + * ACPI/PCI data is missing or mapped differently. + */ +struct sof_dev_desc { + /* list of machines using this configuration */ + struct snd_soc_acpi_mach *machines; + + /* Platform resource indexes in BAR / ACPI resources. */ + /* Must set to -1 if not used - add new items to end */ + int resindex_lpe_base; + int resindex_pcicfg_base; + int resindex_imr_base; + int irqindex_host_ipc; + int resindex_dma_base; + + /* DMA only valid when resindex_dma_base != -1*/ + int dma_engine; + int dma_size; + + /* IPC timeouts in ms */ + int ipc_timeout; + int boot_timeout; + + /* defaults for no codec mode */ + const char *nocodec_fw_filename; + const char *nocodec_tplg_filename; +}; + +int sof_nocodec_setup(struct device *dev, + struct snd_sof_pdata *sof_pdata, + struct snd_soc_acpi_mach *mach, + const struct sof_dev_desc *desc, + struct snd_sof_dsp_ops *ops); + +int sof_bes_setup(struct device *dev, struct snd_sof_dsp_ops *ops, + struct snd_soc_dai_link *links, int link_num, + struct snd_soc_card *card); +#endif diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c new file mode 100644 index 000000000000..29458909182a --- /dev/null +++ b/sound/soc/sof/core.c @@ -0,0 +1,373 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2017 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> + */ + +#include <linux/mm.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <asm/page.h> +#include <asm/pgtable.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/sof.h> +#include "sof-priv.h" +#include "ops.h" + +/* SOF defaults if not provided by the platform in ms */ +#define TIMEOUT_DEFAULT_IPC 5 +#define TIMEOUT_DEFAULT_BOOT 100 + +/* + * Generic object lookup APIs. + */ + +struct snd_sof_pcm *snd_sof_find_spcm_dai(struct snd_sof_dev *sdev, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_sof_pcm *spcm = NULL; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (le32_to_cpu(spcm->pcm.dai_id) == rtd->dai_link->id) + return spcm; + } + + return NULL; +} + +struct snd_sof_pcm *snd_sof_find_spcm_name(struct snd_sof_dev *sdev, + char *name) +{ + struct snd_sof_pcm *spcm = NULL; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (strcmp(spcm->pcm.dai_name, name) == 0) + return spcm; + + if (strcmp(spcm->pcm.caps[0].name, name) == 0) + return spcm; + + if (strcmp(spcm->pcm.caps[1].name, name) == 0) + return spcm; + } + + return NULL; +} + +struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_sof_dev *sdev, + unsigned int comp_id, + int *direction) +{ + struct snd_sof_pcm *spcm = NULL; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (spcm->stream[SNDRV_PCM_STREAM_PLAYBACK].comp_id == + comp_id) { + *direction = SNDRV_PCM_STREAM_PLAYBACK; + return spcm; + } + if (spcm->stream[SNDRV_PCM_STREAM_CAPTURE].comp_id == comp_id) { + *direction = SNDRV_PCM_STREAM_CAPTURE; + return spcm; + } + } + + return NULL; +} + +struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_sof_dev *sdev, + unsigned int pcm_id) +{ + struct snd_sof_pcm *spcm = NULL; + + list_for_each_entry(spcm, &sdev->pcm_list, list) { + if (le32_to_cpu(spcm->pcm.pcm_id) == pcm_id) + return spcm; + } + + return NULL; +} + +struct snd_sof_widget *snd_sof_find_swidget(struct snd_sof_dev *sdev, + char *name) +{ + struct snd_sof_widget *swidget = NULL; + + list_for_each_entry(swidget, &sdev->widget_list, list) { + if (strcmp(name, swidget->widget->name) == 0) + return swidget; + } + + return NULL; +} + +struct snd_sof_dai *snd_sof_find_dai(struct snd_sof_dev *sdev, + char *name) +{ + struct snd_sof_dai *dai = NULL; + + list_for_each_entry(dai, &sdev->dai_list, list) { + if (!dai->name) + continue; + + if (strcmp(name, dai->name) == 0) + return dai; + } + + return NULL; +} + +static inline unsigned int sof_get_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + +/* + * FW Panic/fault handling. + */ + +struct sof_panic_msg { + u32 id; + const char *msg; +}; + +/* standard FW panic types */ +static const struct sof_panic_msg panic_msg[] = { + {SOF_IPC_PANIC_MEM, "out of memory"}, + {SOF_IPC_PANIC_WORK, "work subsystem init failed"}, + {SOF_IPC_PANIC_IPC, "IPC subsystem init failed"}, + {SOF_IPC_PANIC_ARCH, "arch init failed"}, + {SOF_IPC_PANIC_PLATFORM, "platform init failed"}, + {SOF_IPC_PANIC_TASK, "scheduler init failed"}, + {SOF_IPC_PANIC_EXCEPTION, "runtime exception"}, + {SOF_IPC_PANIC_DEADLOCK, "deadlock"}, + {SOF_IPC_PANIC_STACK, "stack overflow"}, + {SOF_IPC_PANIC_IDLE, "can't enter idle"}, +}; + +int snd_sof_get_status(struct snd_sof_dev *sdev, u32 panic_code, + u32 tracep_code, void *oops, void *stack, + size_t stack_words) +{ + u32 code; + int i; + + /* is firmware dead ? */ + if ((panic_code & SOF_IPC_PANIC_MAGIC_MASK) != SOF_IPC_PANIC_MAGIC) { + dev_err(sdev->dev, "error: unexpected fault 0x%8.8x trace 0x%8.8x\n", + panic_code, tracep_code); + return 0; /* no fault ? */ + } + + code = panic_code & + (SOF_IPC_PANIC_MAGIC_MASK | SOF_IPC_PANIC_CODE_MASK); + + for (i = 0; i < ARRAY_SIZE(panic_msg); i++) { + if (panic_msg[i].id == code) { + dev_err(sdev->dev, "error: %s\n", panic_msg[i].msg); + dev_err(sdev->dev, "error: trace point %8.8x\n", + tracep_code); + goto out; + } + } + + /* unknown error */ + dev_err(sdev->dev, "error: unknown reason %8.8x\n", panic_code); + dev_err(sdev->dev, "error: trace point %8.8x\n", tracep_code); + +out: + sof_oops(sdev, oops); + sof_stack(sdev, oops, stack, stack_words); + return -EFAULT; +} +EXPORT_SYMBOL(snd_sof_get_status); + +/* + * Generic buffer page table creation. + */ + +int snd_sof_create_page_table(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + unsigned char *page_table, size_t size) +{ + int i, pages; + + pages = snd_sgbuf_aligned_pages(size); + + dev_dbg(sdev->dev, "generating page table for %p size 0x%zx pages %d\n", + dmab->area, size, pages); + + for (i = 0; i < pages; i++) { + u32 idx = (((i << 2) + i)) >> 1; + u32 pfn = snd_sgbuf_get_addr(dmab, i * PAGE_SIZE) >> PAGE_SHIFT; + u32 *pg_table; + + dev_dbg(sdev->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn); + + pg_table = (u32 *)(page_table + idx); + + if (i & 1) + *pg_table |= (pfn << 4); + else + *pg_table |= pfn; + } + + return pages; +} + +/* + * SOF Driver enumeration. + */ + +static int sof_probe(struct platform_device *pdev) +{ + struct snd_sof_pdata *plat_data = dev_get_platdata(&pdev->dev); + struct snd_sof_dev *sdev; + int ret; + + sdev = devm_kzalloc(&pdev->dev, sizeof(*sdev), GFP_KERNEL); + if (!sdev) + return -ENOMEM; + + dev_dbg(&pdev->dev, "probing SOF DSP device....\n"); + + /* initialize sof device */ + sdev->dev = &pdev->dev; + sdev->parent = plat_data->dev; + if (plat_data->type == SOF_DEVICE_PCI) + sdev->pci = container_of(plat_data->dev, struct pci_dev, dev); + sdev->ops = plat_data->machine->pdata; + + sdev->pdata = plat_data; + INIT_LIST_HEAD(&sdev->pcm_list); + INIT_LIST_HEAD(&sdev->kcontrol_list); + INIT_LIST_HEAD(&sdev->widget_list); + INIT_LIST_HEAD(&sdev->dai_list); + dev_set_drvdata(&pdev->dev, sdev); + spin_lock_init(&sdev->ipc_lock); + spin_lock_init(&sdev->hw_lock); + + /* set up platform component driver */ + snd_sof_new_platform_drv(sdev); + + /* set default timeouts if none provided */ + if (plat_data->desc->ipc_timeout == 0) + sdev->ipc_timeout = TIMEOUT_DEFAULT_IPC; + else + sdev->ipc_timeout = plat_data->desc->ipc_timeout; + if (plat_data->desc->boot_timeout == 0) + sdev->boot_timeout = TIMEOUT_DEFAULT_BOOT; + else + sdev->boot_timeout = plat_data->desc->boot_timeout; + + /* probe the DSP hardware */ + ret = snd_sof_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to probe DSP %d\n", ret); + return ret; + } + + /* register any debug/trace capabilities */ + ret = snd_sof_dbg_init(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to init DSP trace/debug %d\n", + ret); + goto dbg_err; + } + + /* init the IPC */ + sdev->ipc = snd_sof_ipc_init(sdev); + if (!sdev->ipc) { + dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret); + goto ipc_err; + } + + /* load the firmware */ + ret = snd_sof_load_firmware(sdev, plat_data->fw); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to load DSP firmware %d\n", + ret); + goto fw_load_err; + } + + /* boot the firmware */ + ret = snd_sof_run_firmware(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: failed to boot DSP firmware %d\n", + ret); + goto fw_run_err; + } + + /* now register audio DSP platform driver and dai */ + ret = snd_soc_register_component(&pdev->dev, &sdev->plat_drv, + sdev->ops->drv, + sdev->ops->num_drv); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to register DSP DAI driver %d\n", ret); + goto comp_err; + } + + /* init DMA trace */ + ret = snd_sof_init_trace(sdev); + if (ret < 0) { + /* non fatal */ + dev_warn(sdev->dev, + "warning: failed to initialize trace %d\n", ret); + } + + return 0; + +comp_err: + snd_soc_unregister_component(&pdev->dev); + snd_sof_free_topology(sdev); +fw_run_err: + snd_sof_fw_unload(sdev); +fw_load_err: + snd_sof_ipc_free(sdev); +ipc_err: + snd_sof_free_debug(sdev); +dbg_err: + snd_sof_remove(sdev); + + return ret; +} + +static int sof_remove(struct platform_device *pdev) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_component(&pdev->dev); + snd_sof_fw_unload(sdev); + snd_sof_ipc_free(sdev); + snd_sof_free_debug(sdev); + snd_sof_release_trace(sdev); + snd_sof_remove(sdev); + return 0; +} + +void snd_sof_shutdown(struct device *dev) +{ +} +EXPORT_SYMBOL(snd_sof_shutdown); + +static struct platform_driver sof_driver = { + .driver = { + .name = "sof-audio", + }, + + .probe = sof_probe, + .remove = sof_remove, +}; +module_platform_driver(sof_driver); + +MODULE_AUTHOR("Liam Girdwood"); +MODULE_DESCRIPTION("Sound Open Firmware (SOF) Core"); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_ALIAS("platform:sof-audio"); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h new file mode 100644 index 000000000000..adce6c4dff87 --- /dev/null +++ b/sound/soc/sof/sof-priv.h @@ -0,0 +1,504 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2017 Intel Corporation. All rights reserved. + * + * Author: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> + */ + +#ifndef __SOUND_SOC_SOF_PRIV_H +#define __SOUND_SOC_SOF_PRIV_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <uapi/sound/sof-ipc.h> +#include <uapi/sound/sof-fw.h> +#include <uapi/sound/asoc.h> +#include <sound/hdaudio.h> +#include <sound/compress_driver.h> + +/* debug flags */ +#define SOF_DBG_REGS BIT(1) +#define SOF_DBG_MBOX BIT(2) +#define SOF_DBG_TEXT BIT(3) +#define SOF_DBG_PCI BIT(4) + +/* max BARs mmaped devices can use */ +#define SND_SOF_BARS 8 + +/* time in ms for runtime suspend delay */ +#define SND_SOF_SUSPEND_DELAY 2000 + +/* DMA buffer size for trace */ +#define DMA_BUF_SIZE_FOR_TRACE (PAGE_SIZE * 16) + +/* max number of FE PCMs before BEs */ +#define SOF_BE_PCM_BASE 16 + +#define SOF_IPC_DSP_REPLY 0 +#define SOF_IPC_HOST_REPLY 1 + +/* convenience constructor for DAI driver streams */ +#define SOF_DAI_STREAM(sname, scmin, scmax, srates, sfmt) \ + {.stream_name = sname, .channels_min = scmin, .channels_max = scmax, \ + .rates = srates, .formats = sfmt} + +#define SOF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT) + +struct snd_sof_dev; +struct snd_sof_ipc_msg; +struct snd_sof_ipc; +struct snd_sof_debugfs_map; +struct snd_soc_tplg_ops; +struct snd_soc_component; +struct sof_intel_hda_dev; +struct snd_sof_pdata; + +/* + * SOF DSP HW abstraction operations. + * Used to abstract DSP HW architecture and any IO busses between host CPU + * and DSP device(s). + */ +struct snd_sof_dsp_ops { + /* probe and remove */ + int (*remove)(struct snd_sof_dev *sof_dev); + int (*probe)(struct snd_sof_dev *sof_dev); + + /* DSP core boot / reset */ + int (*run)(struct snd_sof_dev *sof_dev); + int (*stall)(struct snd_sof_dev *sof_dev); + int (*reset)(struct snd_sof_dev *sof_dev); + + /* DSP PM */ + int (*suspend)(struct snd_sof_dev *sof_dev, int state); + int (*resume)(struct snd_sof_dev *sof_dev); + + /* DSP clocking */ + int (*set_clk)(struct snd_sof_dev *sof_dev, u32 freq); + + /* Register IO */ + void (*write)(struct snd_sof_dev *sof_dev, void __iomem *addr, + u32 value); + u32 (*read)(struct snd_sof_dev *sof_dev, void __iomem *addr); + void (*write64)(struct snd_sof_dev *sof_dev, void __iomem *addr, + u64 value); + u64 (*read64)(struct snd_sof_dev *sof_dev, void __iomem *addr); + + /* memcpy IO */ + void (*block_read)(struct snd_sof_dev *sof_dev, + u32 offset, void *dest, size_t size); + void (*block_write)(struct snd_sof_dev *sof_dev, + u32 offset, void *src, size_t size); + + /* doorbell */ + irqreturn_t (*irq_handler)(int irq, void *context); + irqreturn_t (*irq_thread)(int irq, void *context); + + /* mailbox */ + void (*mailbox_read)(struct snd_sof_dev *sof_dev, u32 offset, + void *addr, size_t bytes); + void (*mailbox_write)(struct snd_sof_dev *sof_dev, u32 offset, + void *addr, size_t bytes); + + /* ipc */ + int (*send_msg)(struct snd_sof_dev *sof_dev, + struct snd_sof_ipc_msg *msg); + int (*get_reply)(struct snd_sof_dev *sof_dev, + struct snd_sof_ipc_msg *msg); + int (*is_ready)(struct snd_sof_dev *sof_dev); + int (*cmd_done)(struct snd_sof_dev *sof_dev, int dir); + + /* debug */ + const struct snd_sof_debugfs_map *debug_map; + int debug_map_count; + void (*dbg_dump)(struct snd_sof_dev *sof_dev, u32 flags); + + /* connect pcm substream to a host stream */ + int (*pcm_open)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); + /* disconnect pcm substream to a host stream */ + int (*pcm_close)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream); + + /* host stream hw params */ + int (*pcm_hw_params)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); + + /* host stream trigger */ + int (*pcm_trigger)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, int cmd); + + /* FW loading */ + int (*load_firmware)(struct snd_sof_dev *sof_dev, + const struct firmware *fw); + int (*load_module)(struct snd_sof_dev *sof_dev, + struct snd_sof_mod_hdr *hdr); + int (*fw_ready)(struct snd_sof_dev *sdev, u32 msg_id); + + /* host DMA trace initialization */ + int (*trace_init)(struct snd_sof_dev *sdev, u32 *stream_tag); + int (*trace_release)(struct snd_sof_dev *sdev); + int (*trace_trigger)(struct snd_sof_dev *sdev, int cmd); + + /* DAI ops */ + struct snd_soc_dai_driver *drv; + int num_drv; +}; + +/* DSP architecture specific callbacks for oops and stack dumps */ +struct sof_arch_ops { + void (*dsp_oops)(struct snd_sof_dev *sdev, void *oops); + void (*dsp_stack)(struct snd_sof_dev *sdev, void *oops, + u32 *stack, u32 stack_words); +}; + +/* DSP device HW descriptor mapping between bus ID and ops */ +struct sof_ops_table { + const struct sof_dev_desc *desc; + struct snd_sof_dsp_ops *ops; + struct platform_device *(*new_data)(struct snd_sof_pdata *pdata); +}; + +/* FS entry for debug files that can expose DSP memories, registers */ +struct snd_sof_dfsentry_io { + struct dentry *dfsentry; + size_t size; + void __iomem *buf; + struct snd_sof_dev *sdev; +}; + +struct snd_sof_dfsentry_buf { + struct dentry *dfsentry; + size_t size; + void *buf; + struct snd_sof_dev *sdev; +}; + +/* Debug mapping for any DSP memory or registers that can used for debug */ +struct snd_sof_debugfs_map { + const char *name; + u32 bar; + u32 offset; + u32 size; +}; + +/* mailbox descriptor, used for host <-> DSP IPC */ +struct snd_sof_mailbox { + u32 offset; + size_t size; +}; + +/* IPC message descriptor for host <-> DSP IO */ +struct snd_sof_ipc_msg { + struct list_head list; + + /* message data */ + u32 header; + void *msg_data; + void *reply_data; + size_t msg_size; + size_t reply_size; + + wait_queue_head_t waitq; + bool complete; +}; + +/* PCM stream, mapped to FW component */ +struct snd_sof_pcm_stream { + u32 comp_id; + struct snd_dma_buffer page_table; + struct sof_ipc_stream_posn posn; + struct snd_pcm_substream *substream; +}; + +/* ASLA SOF PCM device */ +struct snd_sof_pcm { + struct snd_sof_dev *sdev; + struct snd_soc_tplg_pcm pcm; + struct snd_sof_pcm_stream stream[2]; + u32 posn_offset[2]; + struct mutex mutex; /* access mutex */ + struct list_head list; /* list in sdev pcm list */ +}; + +/* ALSA SOF Kcontrol device */ +struct snd_sof_control { + struct snd_sof_dev *sdev; + int comp_id; + int num_channels; + u32 readback_offset; /* offset to mmaped data if used */ + struct sof_ipc_ctrl_data *control_data; + u32 size; /* cdata size */ + enum sof_ipc_ctrl_cmd cmd; + u32 *volume_table; /* volume table computed from tlv data*/ + + struct mutex mutex; /* access mutex */ + struct list_head list; /* list in sdev control list */ +}; + +/* ASoC SOF DAPM widget */ +struct snd_sof_widget { + struct snd_sof_dev *sdev; + int comp_id; + int pipeline_id; + int complete; + int id; + + struct snd_soc_dapm_widget *widget; + struct mutex mutex; /* access mutex */ + struct list_head list; /* list in sdev widget list */ + + void *private; /* core does not touch this */ +}; + +/* ASoC DAI device */ +struct snd_sof_dai { + struct snd_sof_dev *sdev; + const char *name; + + struct sof_ipc_comp_dai comp_dai; + struct sof_ipc_dai_config dai_config; + struct list_head list; /* list in sdev dai list */ +}; + +/* + * SOF Device Level. + */ +struct snd_sof_dev { + struct device *dev; + struct device *parent; + spinlock_t ipc_lock; /* lock for IPC users */ + spinlock_t hw_lock; /* lock for HW IO access */ + struct pci_dev *pci; + + /* ASoC components */ + struct snd_soc_component_driver plat_drv; + + /* DSP firmware boot */ + wait_queue_head_t boot_wait; + bool boot_complete; + + /* DSP HW differentiation */ + struct snd_sof_pdata *pdata; + const struct snd_sof_dsp_ops *ops; + struct sof_intel_hda_dev *hda; /* for HDA based DSP HW */ + const struct sof_arch_ops *arch_ops; + + /* IPC */ + struct snd_sof_ipc *ipc; + struct snd_sof_mailbox dsp_box; /* DSP initiated IPC */ + struct snd_sof_mailbox host_box; /* Host initiated IPC */ + struct snd_sof_mailbox stream_box; /* Stream position update */ + u64 irq_status; + int ipc_irq; + u32 next_comp_id; /* monotonic - reset during S3 */ + + /* memory bases for mmaped DSPs - set by dsp_init() */ + void __iomem *bar[SND_SOF_BARS]; /* DSP base address */ + int mmio_bar; + int mailbox_bar; + size_t dsp_oops_offset; + + /* debug */ + struct dentry *debugfs_root; + + /* firmware loader */ + int cl_bar; + struct snd_dma_buffer dmab; + struct sof_ipc_fw_ready fw_ready; + + /* topology */ + struct snd_soc_tplg_ops *tplg_ops; + struct list_head pcm_list; + struct list_head kcontrol_list; + struct list_head widget_list; + struct list_head dai_list; + struct snd_soc_component *component; + + /* FW configuration */ + struct sof_ipc_dma_buffer_data *info_buffer; + struct sof_ipc_window *info_window; + + /* IPC timeouts in ms */ + int ipc_timeout; + int boot_timeout; + + /* Wait queue for code loading */ + wait_queue_head_t waitq; + int code_loading; + + /* DMA for Trace */ + struct snd_dma_buffer dmatb; + struct snd_dma_buffer dmatp; + int dma_trace_pages; + wait_queue_head_t trace_sleep; + u32 host_offset; + bool dtrace_is_enabled; + bool dtrace_error; + + void *private; /* core does not touch this */ +}; + +/* + * SOF platform private struct used as drvdata of + * platform dev (e.g. pci/acpi/spi...) drvdata. + */ +struct sof_platform_priv { + struct snd_sof_pdata *sof_pdata; + struct platform_device *pdev_pcm; +}; + +/* + * Device Level. + */ +void snd_sof_shutdown(struct device *dev); +int snd_sof_runtime_suspend(struct device *dev); +int snd_sof_runtime_resume(struct device *dev); +int snd_sof_resume(struct device *dev); +int snd_sof_suspend(struct device *dev); +int snd_sof_suspend_late(struct device *dev); + +void snd_sof_new_platform_drv(struct snd_sof_dev *sdev); +void snd_sof_new_dai_drv(struct snd_sof_dev *sdev); + +int snd_sof_create_page_table(struct snd_sof_dev *sdev, + struct snd_dma_buffer *dmab, + unsigned char *page_table, size_t size); + +/* + * Firmware loading. + */ +int snd_sof_load_firmware(struct snd_sof_dev *sdev, + const struct firmware *fw); +int snd_sof_load_firmware_memcpy(struct snd_sof_dev *sdev, + const struct firmware *fw); +int snd_sof_run_firmware(struct snd_sof_dev *sdev); +int snd_sof_parse_module_memcpy(struct snd_sof_dev *sdev, + struct snd_sof_mod_hdr *module); +void snd_sof_fw_unload(struct snd_sof_dev *sdev); +int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 offset); + +/* + * IPC low level APIs. + */ +struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev); +void snd_sof_ipc_free(struct snd_sof_dev *sdev); +int snd_sof_ipc_reply(struct snd_sof_dev *sdev, u32 msg_id); +void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev); +void snd_sof_ipc_msgs_tx(struct snd_sof_dev *sdev); +int snd_sof_ipc_stream_pcm_params(struct snd_sof_dev *sdev, + struct sof_ipc_pcm_params *params); +int snd_sof_dsp_mailbox_init(struct snd_sof_dev *sdev, u32 dspbox, + size_t dspbox_size, u32 hostbox, + size_t hostbox_size); +int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, void *tx_data, + size_t tx_bytes, void *rx_data, size_t rx_bytes); +struct snd_sof_widget *snd_sof_find_swidget(struct snd_sof_dev *sdev, + char *name); +struct snd_sof_dai *snd_sof_find_dai(struct snd_sof_dev *sdev, + char *name); +struct snd_sof_pcm *snd_sof_find_spcm_dai(struct snd_sof_dev *sdev, + struct snd_soc_pcm_runtime *rtd); +struct snd_sof_pcm *snd_sof_find_spcm_name(struct snd_sof_dev *sdev, + char *name); +struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_sof_dev *sdev, + unsigned int comp_id, + int *direction); +struct snd_sof_pcm *snd_sof_find_spcm_pcm_id(struct snd_sof_dev *sdev, + unsigned int pcm_id); +void sof_ipc_drop_all(struct snd_sof_ipc *ipc); + +/* + * Stream IPC + */ +int snd_sof_ipc_stream_posn(struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm, int direction, + struct sof_ipc_stream_posn *posn); + +/* + * Mixer IPC + */ +int snd_sof_ipc_set_comp_data(struct snd_sof_ipc *ipc, + struct snd_sof_control *scontrol, u32 ipc_cmd, + enum sof_ipc_ctrl_type ctrl_type, + enum sof_ipc_ctrl_cmd ctrl_cmd); +int snd_sof_ipc_get_comp_data(struct snd_sof_ipc *ipc, + struct snd_sof_control *scontrol, u32 ipc_cmd, + enum sof_ipc_ctrl_type ctrl_type, + enum sof_ipc_ctrl_cmd ctrl_cmd); + +/* + * Topology. + */ +int snd_sof_init_topology(struct snd_sof_dev *sdev, + struct snd_soc_tplg_ops *ops); +int snd_sof_load_topology(struct snd_sof_dev *sdev, const char *file); +void snd_sof_free_topology(struct snd_sof_dev *sdev); + +/* + * Trace/debug + */ +int snd_sof_init_trace(struct snd_sof_dev *sdev); +void snd_sof_release_trace(struct snd_sof_dev *sdev); +int snd_sof_dbg_init(struct snd_sof_dev *sdev); +void snd_sof_free_debug(struct snd_sof_dev *sdev); +int snd_sof_debugfs_create_item(struct snd_sof_dev *sdev, + void __iomem *base, size_t size, + const char *name); +int snd_sof_trace_update_pos(struct snd_sof_dev *sdev, + struct sof_ipc_dma_trace_posn *posn); +void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev); +int snd_sof_get_status(struct snd_sof_dev *sdev, u32 panic_code, + u32 tracep_code, void *oops, void *stack, + size_t stack_size); + +/* + * Platform specific ops. + */ +extern struct snd_compr_ops sof_compressed_ops; + +/* + * Kcontrols. + */ + +int snd_sof_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_sof_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + +/* + * DSP Architectures. + */ +static inline void sof_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack, + u32 stack_words) +{ + if (sdev->arch_ops->dsp_stack) + sdev->arch_ops->dsp_stack(sdev, oops, stack, stack_words); +} + +static inline void sof_oops(struct snd_sof_dev *sdev, void *oops) +{ + if (sdev->arch_ops->dsp_oops) + sdev->arch_ops->dsp_oops(sdev, oops); +} + +extern const struct sof_arch_ops sof_xtensa_arch_ops; +#endif -- 2.17.1

4 22

[alsa-devel] [PATCH v2] ALSA: pcm: Return 0 when size < start_threshold in capture
by Ricardo Biehl Pasquali 22 Aug '18

22 Aug '18

In __snd_pcm_lib_xfer(), when capture, if state is PREPARED and size is less than start_threshold nothing can be done. As there is no error, 0 is returned. Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- version 2: The code was not changed. Just resubmitting to follow the rules from Documentation/process/submitting-patches.rst. --- pcm_lib.c.orig 2018-08-21 15:31:54.457405053 -0300 +++ pcm_lib.c 2018-08-21 15:56:01.264549084 -0300 @@ -2125,7 +2125,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(str pcm_transfer_f transfer; bool nonblock; bool is_playback; - int err; + int err = 0; err = pcm_sanity_check(substream); if (err < 0) @@ -2173,11 +2173,15 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(str goto _end_unlock; if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED && - size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) + runtime->status->state == SNDRV_PCM_STATE_PREPARED) { + if (size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) + goto _end_unlock; + } else { + /* nothing to do */ goto _end_unlock; + } } runtime->twake = runtime->control->avail_min ? : 1;

2 3

[alsa-devel] [PATCH] ASoC: sigmadsp: safeload should be used from 4 bytes
by Robert Rosengren 22 Aug '18

22 Aug '18

From: Danny Smith <dannys(a)axis.com> Fixed range in safeload conditional to allow safeload to be used from 4 bytes. Signed-off-by: Danny Smith <dannys(a)axis.com> --- sound/soc/codecs/sigmadsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index d53680ac78e4..d63d58350cd8 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -117,7 +117,7 @@ static int sigmadsp_ctrl_write(struct sigmadsp *sigmadsp, struct sigmadsp_control *ctrl, void *data) { /* safeload loads up to 20 bytes in a atomic operation */ - if (ctrl->num_bytes > 4 && ctrl->num_bytes <= 20 && sigmadsp->ops && + if (ctrl->num_bytes >= 4 && ctrl->num_bytes <= 20 && sigmadsp->ops && sigmadsp->ops->safeload) return sigmadsp->ops->safeload(sigmadsp, ctrl->addr, data, ctrl->num_bytes); -- 2.11.0

3 5

[alsa-devel] [PATCH] ALSA: pcm: Return 0 when size < start_threshold in capture
by Ricardo Biehl Pasquali 22 Aug '18

22 Aug '18

In __snd_pcm_lib_xfer(), when capture, if state is PREPARED and size is less than start_threshold nothing can be done. As there is no error, 0 is returned. Am I missing something? Cheers! pasquali --- pcm_lib.c.orig 2018-08-21 15:31:54.457405053 -0300 +++ pcm_lib.c 2018-08-21 15:56:01.264549084 -0300 @@ -2125,7 +2125,7 @@ pcm_transfer_f transfer; bool nonblock; bool is_playback; - int err; + int err = 0; err = pcm_sanity_check(substream); if (err < 0) @@ -2173,11 +2173,15 @@ goto _end_unlock; if (!is_playback && - runtime->status->state == SNDRV_PCM_STATE_PREPARED && - size >= runtime->start_threshold) { - err = snd_pcm_start(substream); - if (err < 0) + runtime->status->state == SNDRV_PCM_STATE_PREPARED) { + if (size >= runtime->start_threshold) { + err = snd_pcm_start(substream); + if (err < 0) + goto _end_unlock; + } else { + /* nothing to do */ goto _end_unlock; + } } runtime->twake = runtime->control->avail_min ? : 1;

2 1

[alsa-devel] ASOC driver I2C communication
by Riccardo Casagrande 22 Aug '18

22 Aug '18

Hello all, I've developed a driver+codec for a custom sound card. The problem is: the driver take control of the device (The device is marked UU in i2cdetect command) and do not allow to communicate in the userspace with it via I2C. The device results to be always busy. I need to send some commands to the audio chip. Is there a way to use I2C in the userspace? Thanks in advance. Riccardo

1 0

[alsa-devel] [PATCH v3] ASoC: adau17x1: Implemented safeload support
by Robert Rosengren 21 Aug '18

21 Aug '18

From: Danny Smith <dannys(a)axis.com> Safeload support has been implemented which is used when updating for instance filter parameters using alsa controls. Without safeload support audio can become distorted during update. Signed-off-by: Danny Smith <dannys(a)axis.com> Signed-off-by: Robert Rosengren <robertr(a)axis.com> --- sound/soc/codecs/adau17x1.c | 77 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 75 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 2f2afb4c0c88..3959e6ad113d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -21,11 +21,18 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> #include <linux/regmap.h> +#include <asm/unaligned.h> #include "sigmadsp.h" #include "adau17x1.h" #include "adau-utils.h" +#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006 +#define ADAU17X1_SAFELOAD_TRIGGER 0x0007 +#define ADAU17X1_SAFELOAD_DATA 0x0001 +#define ADAU17X1_SAFELOAD_DATA_SIZE 20 +#define ADAU17X1_WORD_SIZE 4 + static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", }; @@ -328,6 +335,17 @@ static bool adau17x1_has_dsp(struct adau *adau) } } +static bool adau17x1_has_safeload(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1781: + return true; + default: + return false; + } +} + static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { @@ -958,6 +976,56 @@ int adau17x1_resume(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(adau17x1_resume); +static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + uint8_t buf[ADAU17X1_WORD_SIZE]; + uint8_t data[ADAU17X1_SAFELOAD_DATA_SIZE]; + unsigned int addr_offset; + unsigned int nbr_words; + int ret; + + /* write data to safeload addresses. Check if len is not a multiple of + * 4 bytes, if so we need to zero pad. + */ + nbr_words = len / ADAU17X1_WORD_SIZE; + if ((len - nbr_words * ADAU17X1_WORD_SIZE) == 0) { + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, bytes, len); + } else { + nbr_words++; + memset(data, 0, ADAU17X1_SAFELOAD_DATA_SIZE); + memcpy(data, bytes, len); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA, data, + nbr_words * ADAU17X1_WORD_SIZE); + } + + if (ret < 0) + return ret; + + /* Write target address, target address is offset by 1 */ + addr_offset = addr - 1; + put_unaligned_be32(addr_offset, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + /* write nbr of words to trigger address */ + put_unaligned_be32(nbr_words, buf); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TRIGGER, buf, ADAU17X1_WORD_SIZE); + if (ret < 0) + return ret; + + return 0; +} + +static const struct sigmadsp_ops adau17x1_sigmadsp_ops = { + .safeload = adau17x1_safeload, +}; + int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name) @@ -1003,8 +1071,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); if (firmware_name) { - adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, - firmware_name); + if (adau17x1_has_safeload(adau)) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + &adau17x1_sigmadsp_ops, firmware_name); + } else { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + NULL, firmware_name); + } if (IS_ERR(adau->sigmadsp)) { dev_warn(dev, "Could not find firmware file: %ld\n", PTR_ERR(adau->sigmadsp)); -- 2.11.0

2 1

[alsa-devel] Patch for HP workstation
by Kailang 21 Aug '18

21 Aug '18

Hi Takashi, Attach patch was fixed for Headset Mic can't recording. BR, Kailang

3 2

[alsa-devel] [PATCH] ALSA: ac97: fix device initialization in the compat layer
by ylhuajnu＠163.com 20 Aug '18

20 Aug '18

From: Lihua Yao <ylhuajnu(a)163.com> ac97->dev is an object of 'struct device' type. It should be initialized via device_initialize() or device_register(). Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus") Signed-off-by: Lihua Yao <ylhuajnu(a)163.com> --- sound/ac97/snd_ac97_compat.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/ac97/snd_ac97_compat.c b/sound/ac97/snd_ac97_compat.c index 61544e0d8de4..8bab44f74bb8 100644 --- a/sound/ac97/snd_ac97_compat.c +++ b/sound/ac97/snd_ac97_compat.c @@ -15,6 +15,11 @@ #include "ac97_core.h" +static void compat_ac97_release(struct device *dev) +{ + kfree(to_ac97_t(dev)); +} + static void compat_ac97_reset(struct snd_ac97 *ac97) { struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); @@ -65,21 +70,31 @@ static struct snd_ac97_bus compat_soc_ac97_bus = { struct snd_ac97 *snd_ac97_compat_alloc(struct ac97_codec_device *adev) { struct snd_ac97 *ac97; + int ret; ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); if (ac97 == NULL) return ERR_PTR(-ENOMEM); - ac97->dev = adev->dev; ac97->private_data = adev; ac97->bus = &compat_soc_ac97_bus; + + ac97->dev.parent = &adev->dev; + ac97->dev.release = compat_ac97_release; + dev_set_name(&ac97->dev, "%s-compat", dev_name(&adev->dev)); + ret = device_register(&ac97->dev); + if (ret) { + put_device(&ac97->dev); + return ERR_PTR(ret); + } + return ac97; } EXPORT_SYMBOL_GPL(snd_ac97_compat_alloc); void snd_ac97_compat_release(struct snd_ac97 *ac97) { - kfree(ac97); + device_unregister(&ac97->dev); } EXPORT_SYMBOL_GPL(snd_ac97_compat_release); -- 2.17.1

5 5

[alsa-devel] Applied "ASoC: rt5514: Fix the issue of the delay volume applied again" to the asoc tree
by Mark Brown 20 Aug '18

20 Aug '18

The patch ASoC: rt5514: Fix the issue of the delay volume applied again has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6f0a256253f48095ba2e5bcdfbed41f21643c105 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 15 Aug 2018 14:47:49 +0800 Subject: [PATCH] ASoC: rt5514: Fix the issue of the delay volume applied again After our evaluation, we need to modify the default values to make sure the volume applied immediately. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index dca82dd6e3bf..32fe76c3134a 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.18.0

1 0

[alsa-devel] [PATCH] slimbus: ngd: mark PM functions as __maybe_unused
by Arnd Bergmann 20 Aug '18

20 Aug '18

qcom_slim_ngd_runtime_suspend is protected by an #ifdef, qcom_slim_ngd_runtime_idle is now, which causes a build time warning: drivers/slimbus/qcom-ngd-ctrl.c:1470:12: error: 'qcom_slim_ngd_runtime_idle' defined but not used [-Werror=unused-function] Marking both as __maybe_unused lets us get rid of the warning as well as the #ifdef. Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/slimbus/qcom-ngd-ctrl.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 8be4d6786c61..14a9d18306cb 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1467,7 +1467,7 @@ static int qcom_slim_ngd_remove(struct platform_device *pdev) return 0; } -static int qcom_slim_ngd_runtime_idle(struct device *dev) +static int __maybe_unused qcom_slim_ngd_runtime_idle(struct device *dev) { struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev); @@ -1477,8 +1477,7 @@ static int qcom_slim_ngd_runtime_idle(struct device *dev) return -EAGAIN; } -#ifdef CONFIG_PM -static int qcom_slim_ngd_runtime_suspend(struct device *dev) +static int __maybe_unused qcom_slim_ngd_runtime_suspend(struct device *dev) { struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev); int ret = 0; @@ -1491,7 +1490,6 @@ static int qcom_slim_ngd_runtime_suspend(struct device *dev) return ret; } -#endif static const struct dev_pm_ops qcom_slim_ngd_dev_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, -- 2.18.0

1 0

[alsa-devel] [PATCH v2] ASoC: wm9712: fix replace codec to component
by Marcel Ziswiler 20 Aug '18

20 Aug '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Since commit 143b44845d87 ("ASoC: wm9712: replace codec to component") "wm9712-codec" got renamed to "wm9712-component", however, this change never got propagated down to the actual board/platform drivers. E.g. on Colibri T20 this lead to the following spew upon boot with sound/touch being broken: [ 2.214121] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.222137] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.344384] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.351885] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.668339] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.675811] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.208408] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.216312] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.235397] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.248938] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 14.970443] ALSA device list: [ 14.996628] No soundcards found. This commit finally fixes this again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- Changes in v2: - Simply rename "wm9712-component" back to "wm9712-codec" as suggested by Kuninori and Charles. sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 953d94d50586..ade34c26ad2f 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -719,7 +719,7 @@ static int wm9712_probe(struct platform_device *pdev) static struct platform_driver wm9712_component_driver = { .driver = { - .name = "wm9712-component", + .name = "wm9712-codec", }, .probe = wm9712_probe, -- 2.14.4

4 3

[alsa-devel] revisiting system timer sound application wake up
by Ricardo Biehl Pasquali 20 Aug '18

20 Aug '18

08/08/2018 (ignore this date) PulseAudio tsched ================= Around December 2007 to February 2009 Lennart Poettering has asked some questions in alsa-devel mailing list in order to implement the timer based audio scheduling for PulseAudio. In 2008 he wrote an article named "What's Cooking in PulseAudio's glitch-free Branch". I have not found detailed documentation about PulseAudio timer wake up and I'm not sure about some implementation details. By using the emails from Lennart and his article a Documentation can start to grow. Do PulseAudio synchronize to sound device by adjusting the timer or the amount committed (application pointer)? The nice thing about the second case is that the period time (in application's view) is bonded to system clock instead of sound device's. What timer interface does PulseAudio use? timerfd? POSIX timers? Hardware issues =============== How the deviation of system and sound device clock is today? Perhaps someone that works with sound cards or SoCs can answer this. Do the most of today's hardware has the same clock for sound device and system? How sound devices deal with random accesses to their ring buffer. Do most of them allow writing/reading anywhere and anytime? How is the acurracy and precision of the hardware pointer returned by sound devices? Does SNDRV_PCM_INFO_BATCH flag reveals something about the above one? As PulseAudio runs in many Linux distributions today, it seems an interesting idea to implement an utility to collect some information regarding the above issues. Then make it available to the public. E.g. it would be possible to know the pointer acurracy for specific devices. SCHED_DEADLINE and timerfd ========================== There are a few works over there about SCHED_DEADLINE and audio: - This paper presented in Linux Audio Conference 2011: BAGNOLI, Giacomo; CUCINOTTA, Tommaso; FAGGIOLI, Dario. 2011. Low-Latency Audio on Linux by Means of Real-Time Scheduling. Dario Faggioli is one of the authors of SCHED_DEADLINE. - This presentation of Alessio Balsini: "Experimenting with the Android Audio Pipeline and SCHED_DEADLINE". OSPM 2018. Here is an overview: <https://lwn.net/Articles/754923/>. In SCHED_DEADLINE I assume there is no way to accurately adjust the timer. As well as there is no way to adjust when the first expiration will be. I think this is because of Constant Bandwidth Server algorithm. Must ask to some developer of this. Having this in mind, the only way to synchronize to sound device is by adjusting the amount read/written (application pointer). Timerfd on the other hand is fully adjustable. However, it can be preempted by a high priority task. Anyway I don't see that as a problem as it should ralely happen. Perhaps SCHED_DEADLINE is suitable for professional environment (aiming almost-zero glitches). Thoughts on this? Is there someone planning to implement SCHED_DEADLINE in PulseAudio? I'm currently working on both implementations (not for PulseAudio). See <https://github.com/ricardobiehl/simplesound>. Take a look at 'Documentation/timer_wakeup.rst', 'tools/timer_wakeup.c' and 'tools/deadline_wakeup.c'. 'tools/waveplay.c' play .wav files using the application wakeup method specified in 'tools/Makefile'. Documentation ============= Many of the points I've mentioned should get well covered by some documentation (e.g. ALSA, PulseAudio). Also, I think ALSA's internal (kernel-level) documentation needs to grow. I'm actually working on that, documenting some stuff as I'm developing a small sound library that interacts directly to ALSA in kernel, and the application wakeup using timerfd and SCHED_DEADLINE. Cheers!

1 0

[alsa-devel] [PATCH 1/2] ALSA: ac97: fix check of pm_runtime_get_sync failure
by ylhuajnu＠163.com 19 Aug '18

19 Aug '18

From: Lihua Yao <ylhuajnu(a)163.com> pm_runtime_get_sync returns negative on failure. Fixes: 74426fbff66e ("ALSA: ac97: add an ac97 bus") Signed-off-by: Lihua Yao <ylhuajnu(a)163.com> --- sound/ac97/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 31f858eceffc..c267e44fefa0 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -503,7 +503,7 @@ static int ac97_bus_remove(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret) + if (ret < 0) return ret; ret = adrv->remove(adev); -- 2.17.1

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[alsa-devel] [PATCH v2] control_hw: Fix issue when applying seccomp policy
by Hsin-Yu Chao 19 Aug '18

19 Aug '18

When seccomp policy is applied to filter ioctl syscall with SNDRV_CTL_IOCTL_TLV_COMMAND, SNDRV_CTL_IOCTL_TLV_READ and SNDRV_CTL_IOCTL_TLV_WRITE in whiltelist, alsa-lib still breaks in at snd_ctl_hw_elem_tlv(). The problem behind is because ioctl() takes unsigned long cmd argument, and the signed bit of local int variable could cause 0xff bytes appended after casted to unsigned long. In kernel, seccomp data struct takes 64 bits argument to check against seccomp rules, these unexpected 0xff bytes could make the rule check fail. Fix the problem by passing unsigned int to ioctl. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/control/control_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 68eca522..b54d65f2 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -215,7 +215,7 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, unsigned int numid, unsigned int *tlv, unsigned int tlv_size) { - int inum; + unsigned int inum; snd_ctl_hw_t *hw = handle->private_data; struct snd_ctl_tlv *xtlv; -- 2.18.0.865.gffc8e1a3cd6-goog

2 1

[alsa-devel] Fwd: [PATCH 1/2] ALSA: hda - Turn CX8200 into D3 as well upon reboot
by Ju Hyung Park 19 Aug '18

19 Aug '18

Hi, It's commit d77a4b4a5b0b2ebcbc9840995d91311ef28302ab and f59cf9a0551dd954ad8b752461cf19d9789f4b1d, both which are already merged in Linus's tree. I can't think of a reason why you would have missed this other than the fact that I've used "Cc: <stable(a)vger.kernel.org>" instead of "Cc: stable(a)vger.kernel.org". I doubt it, but would this be the reason? Thanks, On Sat, Aug 18, 2018 at 11:38 PM Greg KH <gregkh(a)linuxfoundation.org> wrote: > > On Sat, Aug 18, 2018 at 11:29:11PM +0900, Ju Hyung Park wrote: > > Hi, Greg. > > > > I'd like to know if there's a reason on why these ALSA patches(2 in > > total) aren't brought to linux-stable yet. > > What specific patches are you referring to here? I do not see anything > in this email :( > > What is the git commit id in Linus's tree? What stable tree(s) do you > want them applied to? > > > I see a lot of patches merged to linux-stable and stable-queue that > > are much newer than those patches. > > > > I'd appreciate it if you can merge those ASAP. > > Have you read: > https://www.kernel.org/doc/html/latest/process/stable-kernel-rules.html > for how to do this properly? > > If not, please do so. > > thanks, > > greg k-h

2 1

[alsa-devel] Fwd: [PATCH 1/2] ALSA: hda - Turn CX8200 into D3 as well upon reboot
by Ju Hyung Park 18 Aug '18

18 Aug '18

Oh, and I forgot to add: this would better be applied to every stable trees, if applicable (probably by checking if the commit f6b28e4ded45bb91bc4cd115d55e35badedfce5f is already merged). Thanks again. On Sat, Aug 18, 2018 at 11:38 PM Greg KH <gregkh(a)linuxfoundation.org> wrote: > > On Sat, Aug 18, 2018 at 11:29:11PM +0900, Ju Hyung Park wrote: > > Hi, Greg. > > > > I'd like to know if there's a reason on why these ALSA patches(2 in > > total) aren't brought to linux-stable yet. > > What specific patches are you referring to here? I do not see anything > in this email :( > > What is the git commit id in Linus's tree? What stable tree(s) do you > want them applied to? > > > I see a lot of patches merged to linux-stable and stable-queue that > > are much newer than those patches. > > > > I'd appreciate it if you can merge those ASAP. > > Have you read: > https://www.kernel.org/doc/html/latest/process/stable-kernel-rules.html > for how to do this properly? > > If not, please do so. > > thanks, > > greg k-h

1 0

[alsa-devel] [PATCH v2] ASoC: tegra: probe deferral error reporting
by Marcel Ziswiler 18 Aug '18

18 Aug '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Actually report the error codes from of_get_named_gpio() resp. devm_gpio_request_one() upon trying to get the codec reset resp. sync GPIOs unless it is just a probe deferral. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- Changes in v2: - Silence probe deferral as suggested by Stefan. - Fix line over 80 characters as reported by checkpatch. sound/soc/tegra/tegra20_ac97.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 682ef33afb5f..2ac6b2d73980 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -351,18 +351,31 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, GPIOF_OUT_INIT_HIGH, "codec-reset"); if (ret) { - dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "could not get codec-reset GPIO: %d\n", + ret); + goto err_clk_put; } } else { - dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); + ret = ac97->reset_gpio; + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "no codec-reset GPIO supplied: %d\n", + ret); + goto err_clk_put; } ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, "nvidia,codec-sync-gpio", 0); if (!gpio_is_valid(ac97->sync_gpio)) { - dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); + ret = ac97->sync_gpio; + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "no codec-sync GPIO supplied: %d\n", + ret); + goto err_clk_put; } -- 2.14.4

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[alsa-devel] hinawa-utils v0.1.0 and libhinawa v1.0.0 release
by Takashi Sakamoto 18 Aug '18

18 Aug '18

Hi, Today I released hinawa-utils v0.1.0[1]. This package consists of Python 3 modules and scripts to control elements such as output volume in Audio and Music units on IEEE 1394 bus, handled by ALSA firewire stack. At present, all of the scripts are designed to work as CUI, without any server applications. They're listed below: * hinawa-alesis-io-cui * CUI tool for Alesis iO|14 and iO|26 * hinawa-bebob-parser * Plug structure parser for BeBoB firmware * hinawa-config-rom-printer * A lexer/parser of configuration ROM on IEEE 1394 bus * hinawa-dg00x-cui * CUI tool for Digidesign Digi 00x family * hinawa-dice-common-cui * CUI tool for Dice common functionalities * hinawa-dice-extension-cui * CUI tool for Dice extended functionalities a.k.a EAP * hinawa-fireworks-cui * CUI tool for Echo Audio Fireworks module * hinawa-focusrite-saffirepro-io-cui * CUI tool for Focusrite SaffirePro IO series * hinawa-griffin-firewave-cui * CUI tool for Griffin Firewave * hinawa-lacie-speakers-cui * CUI tool for Lacie FireWire speakers * hinawa-motu-common-cui * CUI tool for common functionalities of MOTU FireWire series * hinawa-maudio-bebob-cui * CUI tool for M-Audio FireWire series based on BeBoB platform * hinawa-oxfw-generic-cui * CUI tool for OXFW generic functionalities * hinawa-tascam-fireone-cui * CUI tool for Tascam FireOne * hinawa-tascam-fw-rack-cui * CUI tool for rack models of Tascam FireWire series (FW1804) * hinawa-tascam-fw-console-cui * CUI tool for console models of Tascam FireWire series (FW1082/1884) * hinawa-yamaha-terratec-cui * CUI tool for Yamaha GO series and Terratec PHASE series They're written by Python 3 language, with a help of PyGObject[2] for gobject-introspection[3] to libhinawa API. At the same time, I released libhinawa v1.0.0[4]. This I/O library is designed for asynchronous transactions on IEEE 1394 bus via ioctl commands defined by Linux FireWire subsystem. Additionally, this library is also an application of ALSA HwDep interface supported by ALSA firewire stack. An overview of its design is illustrated[5]. I firstly introduced this library three years ago[6]. This v1.0.0 release loses backward compatibility to a part of public APIs against v0 releases in these three years, and now becomes stable with updated major version. Python 3 modules and scripts in hinawa-utils work with libhinawa v1 APIs. If you'd like to use them, please install libhinawa v1 release instead of v0 releases. I note that modules in hinawa-utils are not stable yet, as its version means. This library has already been packaged to Debian distribution[7] and available in it and its derivatives such as Ubuntu. The v1.0.0 release is already available in their repositories[8][9]. For hinawa-utils, I already start packaging to Debian distribution and get a help of Debian contributer, Kentaro Hayashi, again. We already registered ITP[10] and RFS[11] and wait for sponsors of debian developers to publish. If you find bugs, or have wishes, I suggest to file them as issues in each project on github.com. Of course, I welcome your Pull Requests. I note that these projects are heavily based on vendor-dependent functionalities, thus all of filed issues can not be solved because what I can do is for devices I own. # Off the subject, Kentaro is currently on a process to be Debian # Maintainer(DM)[12]. I greatly appreciate his care and contribution to # libhinawa/hinawa-utils. I hope he will get good experiences from his # next great journey as DM. [1] Release version 0.1.0 release · takaswie/hinawa-utils https://github.com/takaswie/hinawa-utils/releases/tag/v0.1.0 [2] Overview — PyGObject https://pygobject.readthedocs.io/en/latest/ [3] Overview — GObject Introspection https://gi.readthedocs.io/en/latest/ [4] Release version 1.0.0 release · takaswie/libhinawa https://github.com/takaswie/libhinawa/releases/tag/1.0.0 [5] libhinawa/doc/overview.png https://github.com/takaswie/libhinawa/blob/master/doc/overview.png [6] [alsa-devel] [RFC][PATCH 00/13] alsa-tools: libhinawa for control applications of FireWire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2015-January/086969.ht… [7] [alsa-devel] About planning of libhinawa ITP to debian project http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/103693.ht… [8] Debian Package Tracker - libhinawa https://tracker.debian.org/pkg/libhinawa [9] libhinawa package : Ubuntu https://launchpad.net/ubuntu/+source/libhinawa [10] #905840 - ITP: hinawa-utils - Utility tools for libhinawa - Debian Bug report logs https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=905840 [11] #905906 - RFS: hinawa-utils/0.0.99-1 [ITP] - Debian Bug report logs https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=905906 [12] Debian New Member Process https://nm.debian.org/process/531 Regards Takashi Sakamoto

1 0

[alsa-devel] [PATCH] sound/core.c: fix wrong return code of rsnd_probe() when devm_kzalloc() fails
by Jiecheng Wu 18 Aug '18

18 Aug '18

Function rsnd_probe() defined in sound/soc/sh/rcar/core.c calls devm_kzalloc(). As devm_kzalloc() may return NULL when OOM happens, the caller should return -ENOMEM when devm_kzalloc() fails. --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index f8425d8..fd92906 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1502,7 +1502,7 @@ static int rsnd_probe(struct platform_device *pdev) */ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); if (!priv) - return -ENODEV; + return -ENOMEM; priv->pdev = pdev; priv->flags = (unsigned long)of_device_get_match_data(dev); -- 2.6.4

1 0

[alsa-devel] [PATCH v6 0/7] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 17 Aug '18

17 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v6 is based on Marks' for-next branch - the merge of topic/drm_audio_component is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: 1. fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) 2. when the screen goes blank, HDMI playback stops. This isn't related to this patchset since we rely on the existing hdac_hdmi codec, but will need additional investigations. Changes v6 (comments from Mark and Takashi) - simplify Kconfig options, remove FORCE for now and make code simpler - change SPDIX style to C++ for C files - add log message if invalid configuration reached - remove static int counter, added private counter in context instead - fixed a couple of indendation issues to make checkpatch.pl --strict happy Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (5): ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 485 ++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 + sound/soc/intel/Kconfig | 9 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 127 +++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 181 +++++++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 ++- sound/soc/intel/skylake/skl.c | 104 +++- sound/soc/intel/skylake/skl.h | 12 +- 17 files changed, 1102 insertions(+), 27 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.17.1

1 8

[alsa-devel] [PATCH v2 1/2] ASoC: adau17x1: Unused exported functions changed to internal
by Robert Rosengren 17 Aug '18

17 Aug '18

adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.11.0

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[alsa-devel] [RFC PATCH] ASoC: core: Optimise suspend/resume of DAPM widgets
by Jon Hunter 16 Aug '18

16 Aug '18

For soundcards that have several DAI links and many DAPM widgets the time taken for snd_soc_suspend to execute has been observed to be several milliseconds. The time is largely spent executing dapm_power_widgets() for each for the DAI links that need to be suspended. Given that dapm_power_widgets() is called again after suspending/resuming the DAI links, one way to optimise the suspend/resume time is to avoid calling dapm_power_widgets() for each DAI link and reduces the suspend time significantly. Please note that this has been observed on the Tegra210 Jetson TX1 platform which is not currently supported in the mainline for audio but has been tested with out-of-tree patches to enable I2S audio. For Tegra210 the audio drivers (for I2S, etc) are implemented as component drivers to offer greater flexibility with dynamic audio routing within the Tegra Audio Processing Engine (APE) from userspace. The downside of this flexibility is that now there are a lof more DAI links and so the time to suspend the soundcard can take several milliseconds. In the resume path, it is not clear if there could be any issues from not sync'ing the DAPM power state until after unmuting and resuming the CPU DAI drivers, because this will happen later with this change. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- It is not 100% clear to me if there are any downsides to this approach but I am looking to find away to avoid calling dapm_power_widgets() so many times in the suspend path when there are many DAI links. Unfortunately, I don't think that I can simply mark some of the DAI links with 'ignore_suspend' because it could be possible that they are active. include/sound/soc-dapm.h | 2 ++ sound/soc/soc-core.c | 18 ++++-------------- sound/soc/soc-dapm.c | 43 ++++++++++++++++++++++++++++++++++++++++--- 3 files changed, 46 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index af9ef16cc34d..5dbd601154c4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -427,6 +427,8 @@ void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event); +void snd_soc_dapm_stream_suspend(struct snd_soc_pcm_runtime *rtd, int stream); +void snd_soc_dapm_stream_resume(struct snd_soc_pcm_runtime *rtd, int stream); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 81b27923303d..03c5fa95afb8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -519,13 +519,8 @@ int snd_soc_suspend(struct device *dev) if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_SUSPEND); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_SUSPEND); + snd_soc_dapm_stream_suspend(rtd, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dapm_stream_suspend(rtd, SNDRV_PCM_STREAM_CAPTURE); } /* Recheck all endpoints too, their state is affected by suspend */ @@ -637,13 +632,8 @@ static void soc_resume_deferred(struct work_struct *work) if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_RESUME); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_RESUME); + snd_soc_dapm_stream_resume(rtd, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dapm_stream_resume(rtd, SNDRV_PCM_STREAM_CAPTURE); } /* unmute any active DACs */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..7d4f75d29d55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4201,7 +4201,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) } static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - int event) + int event, bool power_widgets) { int i; @@ -4209,7 +4209,8 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, for (i = 0; i < rtd->num_codecs; i++) soc_dapm_dai_stream_event(rtd->codec_dais[i], stream, event); - dapm_power_widgets(rtd->card, event); + if (power_widgets) + dapm_power_widgets(rtd->card, event); } /** @@ -4229,7 +4230,43 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, struct snd_soc_card *card = rtd->card; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - soc_dapm_stream_event(rtd, stream, event); + soc_dapm_stream_event(rtd, stream, event, true); + mutex_unlock(&card->dapm_mutex); +} + +/** + * snd_soc_dapm_stream_suspend - send a suspend stream event to the dapm core + * @rtd: PCM runtime data + * @stream: stream name + * + * Sends a suspend stream event to the dapm core. + * + * Returns 0 for success else error. + */ +void snd_soc_dapm_stream_suspend(struct snd_soc_pcm_runtime *rtd, int stream) +{ + struct snd_soc_card *card = rtd->card; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + soc_dapm_stream_event(rtd, stream, SND_SOC_DAPM_STREAM_SUSPEND, false); + mutex_unlock(&card->dapm_mutex); +} + +/** + * snd_soc_dapm_stream_resume - send a resume stream event to the dapm core + * @rtd: PCM runtime data + * @stream: stream name + * + * Sends a resume stream event to the dapm core. + * + * Returns 0 for success else error. + */ +void snd_soc_dapm_stream_resume(struct snd_soc_pcm_runtime *rtd, int stream) +{ + struct snd_soc_card *card = rtd->card; + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + soc_dapm_stream_event(rtd, stream, SND_SOC_DAPM_STREAM_RESUME, false); mutex_unlock(&card->dapm_mutex); } -- 2.7.4

2 6

[alsa-devel] [ANN] Linux Media Summit 2018 together with ELC-EU
by Mauro Carvalho Chehab 16 Aug '18

16 Aug '18

Hi all, We're organizing this year's media summit to happen together with the Embedded Linux Conference - Europe, on Thursday, October 25 2018: https://events.linuxfoundation.org/events/elc-openiot-europe-2018/ Media Summit is the premier forum to discuss the Linux multimedia development for cameras, audio and video streaming devices, analog/digital TV support, remote controller and HDMI Consumer Electronics Control (CEC) at the Linux Kernel and its related userspace APIs. The ones interested on participating on it should be registered at ELC or OSS Europe, and indicate the participation at the media summit via Linux Foundation's register site: https://events.linuxfoundation.org/events/elc-openiot-europe-2018/register/ We're also asking for the themes to be discussed there. Please send your proposals to the linux-media(a)vger.kernel.org mailing list. In order to make easier for us to track, please add a [RFP] at the beginning of the email's subject. Hoping to see you there, Regards, Mauro

1 0

[alsa-devel] [PATCH] ASoC: max98373: Added speaker FS gain cotnrol register to volatile.
by Ryan Lee 16 Aug '18

16 Aug '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Added Speaker FS gain maximum control(0x203E) register to volatile. sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 92b7125..2764fae 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -520,6 +520,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98373_R2000_SW_RESET ... MAX98373_R2009_INT_FLAG3: + case MAX98373_R203E_AMP_PATH_GAIN: case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5514: Fix the issue of the delay volume applied again
by Oder Chiou 16 Aug '18

16 Aug '18

After our evaluation, we need to modify the default values to make sure the volume applied immediately. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index dca82dd..32fe76c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: info: Check for integer overflow in snd_info_entry_write()
by Erick Reyes 16 Aug '18

16 Aug '18

Commit 4adb7bcbcb69 ("ALSA: core: Use seq_file for text proc file reads") heavily refactored ALSA procfs and fixed the overflow as a side-effect, so this fix only applies to kernels < 4.2 and there is no upstream equivalent snd_info_entry_write() resizes the buffer with an unsigned long size argument that gets truncated because resize_info_buffer() takes the size parameter as an unsigned int. On 64-bit kernels, this causes the following copy_to_user() to write out-of-bounds if (pos + count) can't be represented by an unsigned int. Signed-off-by: Siqi Lin <siqilin(a)google.com> Signed-off-by: Erick Reyes <erickreyes(a)google.com> --- sound/core/info.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/info.c b/sound/core/info.c index 9f404e965ea2..08832c973a53 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -253,6 +253,7 @@ static ssize_t snd_info_entry_write(struct file *file, const char __user *buffer struct snd_info_buffer *buf; ssize_t size = 0; loff_t pos; + unsigned long realloc_size; data = file->private_data; if (snd_BUG_ON(!data)) @@ -261,7 +262,8 @@ static ssize_t snd_info_entry_write(struct file *file, const char __user *buffer pos = *offset; if (pos < 0 || (long) pos != pos || (ssize_t) count < 0) return -EIO; - if ((unsigned long) pos + (unsigned long) count < (unsigned long) pos) + realloc_size = (unsigned long) pos + (unsigned long) count; + if (realloc_size < (unsigned long) pos || realloc_size > UINT_MAX) return -EIO; switch (entry->content) { case SNDRV_INFO_CONTENT_TEXT: -- 2.18.0.865.gffc8e1a3cd6-goog

3 2

[alsa-devel] Alsa test application which does both record and render
by Chakravarthi Pradeep 16 Aug '18

16 Aug '18

Dear ALSA Team. I developed ALSA driver which record the audio data from PCI. I would like to test it using unit test application which can record and render frame by frame. Is there any ALSA unit test application which can be used for testing ? Thanks and Regards Chakravarthi Pradeep.K

1 0

[alsa-devel] [PATCH] control_hw: Fix issue when applying seccomp policy
by Hsin-Yu Chao 16 Aug '18

16 Aug '18

When seccomp policy is applied to filter ioctl syscall with SNDRV_CTL_IOCTL_TLV_COMMAND, SNDRV_CTL_IOCTL_TLV_READ and SNDRV_CTL_IOCTL_TLV_WRITE in whiltelist, alsa-lib still breaks in at snd_ctl_hw_elem_tlv(). Fix the problem by passing unsigned int to ioctl. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/control/control_hw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/control_hw.c b/src/control/control_hw.c index 68eca522..b54d65f2 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -215,7 +215,7 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, unsigned int numid, unsigned int *tlv, unsigned int tlv_size) { - int inum; + unsigned int inum; snd_ctl_hw_t *hw = handle->private_data; struct snd_ctl_tlv *xtlv; -- 2.18.0.865.gffc8e1a3cd6-goog

2 3

[alsa-devel] Trying to create a developer account, but need a token
by Gene Cristan 15 Aug '18

15 Aug '18

To help protect against automated account creation, please answer the question that appears below (more info [1]): Token to prevent spam? Please, ask on mailing lists... Links: ------ [1] http://www.alsa-project.org/main/index.php/Special:Captcha/help

1 0

[alsa-devel] [PATCH - hwmixvolume 4/9] hwmixvolume: use a with context to open files
by Emmanuel Gil Peyrot 15 Aug '18

15 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 267228c..e82f32d 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -138,26 +138,20 @@ class Stream: subdevice = info.index filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) try: - f = open(filename, "r") + with open(filename, "r") as f: + for line in f.readlines(): + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) except IOError: return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() return None def get_pid_cmdline(self, pid): try: - f = open("/proc/%d/cmdline" % pid, "r") + with open("/proc/%d/cmdline" % pid, "r") as f: + cmdline = f.read() except IOError: return None - try: - cmdline = f.read() - finally: - f.close() return cmdline.replace('\x00', ' ').strip() class MixerWindow(Gtk.Window): -- 2.18.0

2 2

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 15 Aug '18

15 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs
by Charles Keepax 15 Aug '18

15 Aug '18

Commit a655de808cbde ("ASoC: core: Allow topology to override machine driver FE DAI link config.") caused soc_dai_hw_params to be come dependent on the substream private_data being set with a pointer to the snd_soc_pcm_runtime. Currently, CODEC to CODEC links don't set this, which causes a NULL pointer dereference: [<4069de54>] (soc_dai_hw_params) from [<40694b68>] (snd_soc_dai_link_event+0x1a0/0x380) Since the ASoC core in general assumes that the substream private_data will be set to a pointer to the snd_soc_pcm_runtime, update the CODEC to CODEC links to respect this. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 4 ++++ 3 files changed, 7 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index af9ef16cc34d..fdaaafdc7a00 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -407,6 +407,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, const struct snd_soc_pcm_stream *params, unsigned int num_params, struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9cfe10d8040c..473eefe8658e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1447,7 +1447,7 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, sink = codec_dai->playback_widget; source = cpu_dai->capture_widget; if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, + ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, dai_link->num_params, source, sink); if (ret != 0) { @@ -1460,7 +1460,7 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, sink = cpu_dai->playback_widget; source = codec_dai->capture_widget; if (sink && source) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, + ret = snd_soc_dapm_new_pcm(card, rtd, dai_link->params, dai_link->num_params, source, sink); if (ret != 0) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e96793050c9..461d951917c0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3652,6 +3652,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; + struct snd_soc_pcm_runtime *rtd = w->priv; const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; @@ -3711,6 +3712,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, goto out; } substream.runtime = runtime; + substream.private_data = rtd; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3895,6 +3897,7 @@ snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, const struct snd_soc_pcm_stream *params, unsigned int num_params, struct snd_soc_dapm_widget *source, @@ -3963,6 +3966,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, w->params = params; w->num_params = num_params; + w->priv = rtd; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) -- 2.11.0

2 1

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 15 Aug '18

15 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

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Re: [alsa-devel] ASoC: pxa: possible regressions
by Daniel Mack 15 Aug '18

15 Aug '18

Hi Petr, On Wednesday, August 08, 2018 04:24 AM, Petr Cvek wrote: > I was trying to update my system (PXA Magician) to 4.18-next and i've > ran into few problems with sound. Ah, so you have a Magician platform, that's very good! > My setup has only a few changes from > vanilla (change to using devm_snd_soc_register_card+devm_gpio_request). Care to share these changes, or ideally rebase them on top of 4.18-next for mainline inclusion? Also, I'm curious what mainline version you used to be based on that worked for you. > I've started digging around: Thanks for that. Are you also willing to send patches to fix things once they work for you? > 1) unimportant goto > > Commit 7afd1b0b2ef9a812095 ("ASoC: pxa: move some functions to > pxa2xx-lib") is this construct necessary? > > if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { > ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, > SNDRV_PCM_STREAM_CAPTURE); > if (ret) > goto out; > } > out: > return ret; > > After calling pxa2xx_pcm_preallocate_dma_buffer() the code will jump to > out anyway. True. That's a cosmetic cleanup which can go in for 4.19. Would you want to send a patch after the merge window? > 2) wrong units (sample rate vs osc rate) > > Commit 05739375f1c0a1048 ("ASoC: pxa-ssp: remove .set_pll() and > .set_clkdiv() callbacks"). The function pxa_ssp_hw_params() get's called > with sample rate so: > > /* The values in the table are for 16 bits */ > if (width == 32) > acds--; > > ret = pxa_ssp_set_pll(priv, bclk); > if (ret < 0) > return ret; > > ssacd = pxa_ssp_read_reg(ssp, SSACD); > ssacd &= ~(SSACD_ACDS(7) | SSACD_SCDB_1X); > > pxa_ssp_set_pll() will try to set something like 8000, 44100, ... and > not the speed of an oscilator. The line should be probably: > > ret = pxa_ssp_set_pll(priv, m->pll); > > and there is a previous call to pxa_ssp_set_pll(): Ah, yes. Does changing that fix it for you? The changes in that machine driver where done blindly because I don't have Magician hardware, and I really don't like doing things that way. > 3) setup fails > > Recent changes in ASoC PXA DMA are causing a fail of a condition in > pxa_ssp_hw_params(): > > if ((sscr0 & SSCR0_MOD) && !ttsa) { What are the values of sscr0 and ttsa in your case? Can you cross-check that with a kernel that doesn't have my recent changes? > dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); > return -EINVAL; > } > > It seems the SSCR0_MOD flag is set in the fall through switch statement > in pxa_ssp_configure_dai_fmt(): > > case SND_SOC_DAIFMT_DSP_A: > sspsp |= SSPSP_FSRT; > /* fall through */ > case SND_SOC_DAIFMT_DSP_B: > sscr0 |= SSCR0_MOD | SSCR0_PSP; > sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; > break; > > the switch statement is an old code and it worked in the past versions. Yes, it was merely moved to some location that is called earlier. This was fixing a regression caused by a8bd0ee558714 ("ASoC: raumfeld: Use static DAI format setup"), so supposedly, all PXA sound was broken since v4.0. > 4) names are not unique > > The initialization of magician_dai structure wants to have (at least in > the past versions) for the .name item: > > static struct snd_soc_dai_link magician_dai[] = { > { > .name = "uda1380", > .stream_name = "UDA1380 Playback", > .cpu_dai_name = "pxa-ssp-dai.0", > ... > { > .name = "uda1380", > .stream_name = "UDA1380 Capture", > .cpu_dai_name = "pxa2xx-i2s", > > my personal branch has them named as "uda1380_playback" and > "uda1380_capture". Care to send a patch for that as well, please? > 5) Unbalanced references > > When trying to play a sound (with quick-and-dirty workarounds for > problems above): > > # cat /dev/urandom | aplay - > > the system will crash when terminating aplay (stopping playback). The > backtrace: So you did hear a sound? [...] > I've only managed to discover there are two calls of > pxa2xx_soc_pcm_new() (where DMA buffer is allocated) for both dai links: > > magician-audio magician-audio: uda1380-hifi-playback <-> pxa-ssp-dai.0 > mapping ok > ... > magician-audio magician-audio: uda1380-hifi-capture <-> pxa2xx-i2s > mapping ok > > but I don't know if this information is relevant. That seems fine. The question is why there are are unbalanced calls to snd_dmaengine_pcm_close_release_chan(). Could you dig a bit further maybe and add some printk()? Thanks for helping fix these regressions! Daniel

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[alsa-devel] [PATCH v2 00/10] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 15 Aug '18

15 Aug '18

Thankyou for reviewing v1 patchset, here is v2 addressing comments from v1. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback and capture paths. This patchset also depends on the SLIMbus Stream apis which is already merged via char-misc tree for 4.19. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v1 (https://lkml.org/lkml/2018/7/23/889) - Fixed dt bindings as suggested by Mark B. - cleanup CLASS-H code, remove un-used states. - Various cleanup suggested by Vinod. Srinivas Kandagatla (10): ASoC: dt-bindings: add dt bindings for wcd9335 audio codec mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: add support to Analog audio via WCD9335 slim .../devicetree/bindings/sound/qcom,wcd9335.txt | 123 + drivers/mfd/Kconfig | 18 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 277 ++ drivers/mfd/wcd9335-irq.c | 172 + include/dt-bindings/mfd/wcd9335.h | 43 + include/linux/mfd/wcd9335/registers.h | 580 +++ include/linux/mfd/wcd9335/wcd9335.h | 45 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh.c | 605 +++ sound/soc/codecs/wcd-clsh.h | 49 + sound/soc/codecs/wcd9335.c | 4963 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 68 + 14 files changed, 6954 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd9335.txt create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 drivers/mfd/wcd9335-irq.c create mode 100644 include/dt-bindings/mfd/wcd9335.h create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh.c create mode 100644 sound/soc/codecs/wcd-clsh.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.16.2

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[alsa-devel] Miscellaneous ioplug questions
by Rob Duncan 15 Aug '18

15 Aug '18

I have some questions about some ioplug implementation details: - Is the value returned from the pointer callback supposed to be a monotonically increasing value, or should it wrap at the buffer size? The documentation says "get the current DMA position" which isn't clear to me. - Must the pointer callback return values that are an integral multiple of the period size, or can it report partial periods? - Should the poll FD be made writeable as soon as data is available, or only when at least avail_min is available? Thanks, Rob.

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[alsa-devel] WARNING in __snd_rawmidi_transmit_ack (2)
by syzbot 14 Aug '18

14 Aug '18

Hello, syzbot found the following crash on: HEAD commit: 4e8b38549b50 Add linux-next specific files for 20180814 git tree: linux-next console output: https://syzkaller.appspot.com/x/log.txt?x=11f5e78c400000 kernel config: https://syzkaller.appspot.com/x/.config?x=18edf0289d1b5ab dashboard link: https://syzkaller.appspot.com/bug?extid=ec5f605c91812d200367 compiler: gcc (GCC) 8.0.1 20180413 (experimental) syzkaller repro:https://syzkaller.appspot.com/x/repro.syz?x=12f8c6d2400000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=16976b72400000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+ec5f605c91812d200367(a)syzkaller.appspotmail.com WARNING: CPU: 1 PID: 1691 at sound/core/rawmidi.c:1170 __snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1170 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 1691 Comm: kworker/1:1H Not tainted 4.18.0-next-20180814+ #38 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events_highpri snd_vmidi_output_work Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x1c9/0x2b4 lib/dump_stack.c:113 panic+0x238/0x4e7 kernel/panic.c:184 __warn.cold.8+0x163/0x1ba kernel/panic.c:536 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1fc/0x4d0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:996 RIP: 0010:__snd_rawmidi_transmit_ack+0x263/0x2f0 sound/core/rawmidi.c:1170 Code: ba 01 00 00 00 be 03 00 00 00 e8 18 3a 66 fc e8 e3 44 86 fc 89 d8 48 83 c4 18 5b 41 5c 41 5d 41 5e 41 5f 5d c3 e8 cd 44 86 fc <0f> 0b e9 72 fe ff ff e8 21 5a c5 fc e9 1b ff ff ff bb ea ff ff ff RSP: 0018:ffff8801d2a6f5a8 EFLAGS: 00010093 RAX: ffff8801d2a52300 RBX: 0000000000000001 RCX: ffffffff84f64792 RDX: 0000000000000000 RSI: ffffffff84f64923 RDI: 0000000000000006 RBP: ffff8801d2a6f5e8 R08: ffff8801d2a52300 R09: ffffed003607acc1 R10: ffffed003607acc1 R11: ffff8801b03d660b R12: ffff8801ce2479c0 R13: 0000000000001001 R14: 0000000000000001 R15: ffff8801b03d65c0 snd_rawmidi_transmit_ack+0x51/0x80 sound/core/rawmidi.c:1201 snd_vmidi_output_work+0x3f4/0x4e0 sound/core/seq/seq_virmidi.c:153 process_one_work+0xc73/0x1aa0 kernel/workqueue.c:2153 worker_thread+0x189/0x13c0 kernel/workqueue.c:2296 kthread+0x35a/0x420 kernel/kthread.c:246 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:415 Shutting down cpus with NMI Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#bug-status-tracking for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 14 Aug '18

14 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Acquire irq after RIRB allocation
by Yong Zhi 14 Aug '18

14 Aug '18

Cold reboot stress test found that the hda irq could access rirb ring buffer before its memory gets allocated which resulting in null pointer dereference inside snd_hdac_bus_update_rirb(). Fix it by moving the skl_acquire_irq after ring buffer allocation. While here, also change err return from -EBUSY to actual error code. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/skylake/skl.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index dce649485649..cf09721ca13e 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -838,11 +838,7 @@ static int skl_first_init(struct hdac_bus *bus) snd_hdac_bus_parse_capabilities(bus); - if (skl_acquire_irq(bus, 0) < 0) - return -EBUSY; - pci_set_master(pci); - synchronize_irq(bus->irq); gcap = snd_hdac_chip_readw(bus, GCAP); dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); @@ -875,6 +871,12 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; + err = skl_acquire_irq(bus, 0); + if (err < 0) + return err; + + synchronize_irq(bus->irq); + /* initialize chip */ skl_init_pci(skl); -- 2.7.4

4 4

[alsa-devel] Lenovo Y700-15ISK integrated subwoofer not working
by Leandro Britez 14 Aug '18

14 Aug '18

Hello, I'm having a problem with a Lenovo Y700 (15ISK, with the same hardware configuration as the 17ISK and 14ISK). Integrated frontal speakers and headphone/microphone jack are working fine, but I'm not able to get any sound from the integrated subwoofer, on any distro tested (Ubuntu, Debian, actually using Arch). Googled for solutions but looks like no user using this notebook family has a working solution yet. - I tryied to identify 0x17 and 0x1b pins as LFE or Frontal Speakers with hdajackretask. - Checked with hda-analyzer if some node was muting or blocking that connection and enabled "OUT" on Widget Control for 0x1b pin complex and tested various VREF options. - I also changed POWER_STATE of pretty much everything with hda-verb, with no success. - Checked alsamixer/pulseaudio volumes, mutes, and settings. - Tested some snd-hda-intel model options related to the codec or the notebook manufacturer. I am going to test every model when I get some time. Those pins are the only unidentified ones with Output caps: pin 0x17 <--> aud_mix 0x0f <--> aud_mix 0x0d <--> output 0x03 (headphones) and aud_mix 0x0b (mic playbacks) pin complex 0x1b <--> aud_mix 0x0c <--> output 0x02 (speakers) and aud_mix 0x0b (mic playbacks) 0x1b looks promising and probably the only option, but I'm not able to get it to work. I don't know if I'm missing something or there is something more I can test with that pin complex. Please find the attached alsa-info.sh and lspci -vvnn output files. Thank you for your time. Leandro

1 0

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Add quirks for 2 more devices
by Hans de Goede 14 Aug '18

14 Aug '18

Add quirks to select the right input-map, jack-detect pin, etc. for: Linx Linx7 tablet Onda V975w tablet Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d32844f94d74..b6dc524830b2 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -575,6 +575,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Linx Linx7 tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LINX"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "LINX7"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MSI S100 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."), @@ -602,6 +613,21 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Onda v975w */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* The above are too generic, also match BIOS info */ + DMI_EXACT_MATCH(DMI_BIOS_VERSION, "5.6.5"), + DMI_EXACT_MATCH(DMI_BIOS_DATE, "07/25/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, { /* Pipo W4 */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), -- 2.18.0

3 2

[alsa-devel] Applied "ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors" to the asoc tree
by Mark Brown 14 Aug '18

14 Aug '18

The patch ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fb504caae7ef85be159743bd4b08ecde269ba55f Mon Sep 17 00:00:00 2001 From: "Dmitry V. Levin" <ldv(a)altlinux.org> Date: Mon, 13 Aug 2018 18:50:02 +0300 Subject: [PATCH] ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors Include <linux/types.h> and consistently use types it provides to fix the following sound/skl-tplg-interface.h userspace compilation errors: /usr/include/sound/skl-tplg-interface.h:146:2: error: unknown type name 'u32' u32 set_params:2; /usr/include/sound/skl-tplg-interface.h:147:2: error: unknown type name 'u32' u32 rsvd:30; /usr/include/sound/skl-tplg-interface.h:148:2: error: unknown type name 'u32' u32 param_id; /usr/include/sound/skl-tplg-interface.h:149:2: error: unknown type name 'u32' u32 max; /usr/include/sound/skl-tplg-interface.h:166:2: error: unknown type name 'u16' u16 module_id; /usr/include/sound/skl-tplg-interface.h:167:2: error: unknown type name 'u16' u16 instance_id; /usr/include/sound/skl-tplg-interface.h:171:2: error: unknown type name 'u32' u32 channels; /usr/include/sound/skl-tplg-interface.h:172:2: error: unknown type name 'u32' u32 freq; /usr/include/sound/skl-tplg-interface.h:173:2: error: unknown type name 'u32' u32 bit_depth; /usr/include/sound/skl-tplg-interface.h:174:2: error: unknown type name 'u32' u32 valid_bit_depth; /usr/include/sound/skl-tplg-interface.h:175:2: error: unknown type name 'u32' u32 ch_cfg; /usr/include/sound/skl-tplg-interface.h:176:2: error: unknown type name 'u32' u32 interleaving_style; /usr/include/sound/skl-tplg-interface.h:177:2: error: unknown type name 'u32' u32 sample_type; /usr/include/sound/skl-tplg-interface.h:178:2: error: unknown type name 'u32' u32 ch_map; /usr/include/sound/skl-tplg-interface.h:182:2: error: unknown type name 'u32' u32 set_params:2; /usr/include/sound/skl-tplg-interface.h:183:2: error: unknown type name 'u32' u32 rsvd:30; /usr/include/sound/skl-tplg-interface.h:184:2: error: unknown type name 'u32' u32 param_id; /usr/include/sound/skl-tplg-interface.h:185:2: error: unknown type name 'u32' u32 caps_size; /usr/include/sound/skl-tplg-interface.h:186:2: error: unknown type name 'u32' u32 caps[HDA_SST_CFG_MAX]; /usr/include/sound/skl-tplg-interface.h:190:2: error: unknown type name 'u8' u8 pipe_id; /usr/include/sound/skl-tplg-interface.h:191:2: error: unknown type name 'u8' u8 pipe_priority; /usr/include/sound/skl-tplg-interface.h:192:2: error: unknown type name 'u16' u16 conn_type:4; /usr/include/sound/skl-tplg-interface.h:193:2: error: unknown type name 'u16' u16 rsvd:4; /usr/include/sound/skl-tplg-interface.h:194:2: error: unknown type name 'u16' u16 memory_pages:8; /usr/include/sound/skl-tplg-interface.h:200:2: error: unknown type name 'u16' u16 module_id; /usr/include/sound/skl-tplg-interface.h:201:2: error: unknown type name 'u16' u16 instance_id; /usr/include/sound/skl-tplg-interface.h:202:2: error: unknown type name 'u32' u32 max_mcps; /usr/include/sound/skl-tplg-interface.h:203:2: error: unknown type name 'u32' u32 mem_pages; /usr/include/sound/skl-tplg-interface.h:204:2: error: unknown type name 'u32' u32 obs; /usr/include/sound/skl-tplg-interface.h:205:2: error: unknown type name 'u32' u32 ibs; /usr/include/sound/skl-tplg-interface.h:206:2: error: unknown type name 'u32' u32 vbus_id; /usr/include/sound/skl-tplg-interface.h:208:2: error: unknown type name 'u32' u32 max_in_queue:8; /usr/include/sound/skl-tplg-interface.h:209:2: error: unknown type name 'u32' u32 max_out_queue:8; /usr/include/sound/skl-tplg-interface.h:210:2: error: unknown type name 'u32' u32 time_slot:8; /usr/include/sound/skl-tplg-interface.h:211:2: error: unknown type name 'u32' u32 core_id:4; /usr/include/sound/skl-tplg-interface.h:212:2: error: unknown type name 'u32' u32 rsvd1:4; /usr/include/sound/skl-tplg-interface.h:214:2: error: unknown type name 'u32' u32 module_type:8; /usr/include/sound/skl-tplg-interface.h:215:2: error: unknown type name 'u32' u32 conn_type:4; /usr/include/sound/skl-tplg-interface.h:216:2: error: unknown type name 'u32' u32 dev_type:4; /usr/include/sound/skl-tplg-interface.h:217:2: error: unknown type name 'u32' u32 hw_conn_type:4; /usr/include/sound/skl-tplg-interface.h:218:2: error: unknown type name 'u32' u32 rsvd2:12; /usr/include/sound/skl-tplg-interface.h:220:2: error: unknown type name 'u32' u32 params_fixup:8; /usr/include/sound/skl-tplg-interface.h:221:2: error: unknown type name 'u32' u32 converter:8; /usr/include/sound/skl-tplg-interface.h:222:2: error: unknown type name 'u32' u32 input_pin_type:1; /usr/include/sound/skl-tplg-interface.h:223:2: error: unknown type name 'u32' u32 output_pin_type:1; /usr/include/sound/skl-tplg-interface.h:224:2: error: unknown type name 'u32' u32 is_dynamic_in_pin:1; /usr/include/sound/skl-tplg-interface.h:225:2: error: unknown type name 'u32' u32 is_dynamic_out_pin:1; /usr/include/sound/skl-tplg-interface.h:226:2: error: unknown type name 'u32' u32 is_loadable:1; /usr/include/sound/skl-tplg-interface.h:227:2: error: unknown type name 'u32' u32 rsvd3:11; Fixes: 0c24fdc00244 ("ASoC: topology: Move skl-tplg-interface.h to uapi") Signed-off-by: Dmitry V. Levin <ldv(a)altlinux.org> Reviewed-by: Guenter Roeck <groeck(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: <stable(a)vger.kernel.org> # v4.18 --- include/uapi/sound/skl-tplg-interface.h | 106 ++++++++++++------------ 1 file changed, 54 insertions(+), 52 deletions(-) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index f58cafa42f18..f39352cef382 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -10,6 +10,8 @@ #ifndef __HDA_TPLG_INTERFACE_H__ #define __HDA_TPLG_INTERFACE_H__ +#include <linux/types.h> + /* * Default types range from 0~12. type can range from 0 to 0xff * SST types start at higher to avoid any overlapping in future @@ -143,10 +145,10 @@ enum skl_module_param_type { }; struct skl_dfw_algo_data { - u32 set_params:2; - u32 rsvd:30; - u32 param_id; - u32 max; + __u32 set_params:2; + __u32 rsvd:30; + __u32 param_id; + __u32 max; char params[0]; } __packed; @@ -163,68 +165,68 @@ enum skl_tuple_type { /* v4 configuration data */ struct skl_dfw_v4_module_pin { - u16 module_id; - u16 instance_id; + __u16 module_id; + __u16 instance_id; } __packed; struct skl_dfw_v4_module_fmt { - u32 channels; - u32 freq; - u32 bit_depth; - u32 valid_bit_depth; - u32 ch_cfg; - u32 interleaving_style; - u32 sample_type; - u32 ch_map; + __u32 channels; + __u32 freq; + __u32 bit_depth; + __u32 valid_bit_depth; + __u32 ch_cfg; + __u32 interleaving_style; + __u32 sample_type; + __u32 ch_map; } __packed; struct skl_dfw_v4_module_caps { - u32 set_params:2; - u32 rsvd:30; - u32 param_id; - u32 caps_size; - u32 caps[HDA_SST_CFG_MAX]; + __u32 set_params:2; + __u32 rsvd:30; + __u32 param_id; + __u32 caps_size; + __u32 caps[HDA_SST_CFG_MAX]; } __packed; struct skl_dfw_v4_pipe { - u8 pipe_id; - u8 pipe_priority; - u16 conn_type:4; - u16 rsvd:4; - u16 memory_pages:8; + __u8 pipe_id; + __u8 pipe_priority; + __u16 conn_type:4; + __u16 rsvd:4; + __u16 memory_pages:8; } __packed; struct skl_dfw_v4_module { char uuid[SKL_UUID_STR_SZ]; - u16 module_id; - u16 instance_id; - u32 max_mcps; - u32 mem_pages; - u32 obs; - u32 ibs; - u32 vbus_id; - - u32 max_in_queue:8; - u32 max_out_queue:8; - u32 time_slot:8; - u32 core_id:4; - u32 rsvd1:4; - - u32 module_type:8; - u32 conn_type:4; - u32 dev_type:4; - u32 hw_conn_type:4; - u32 rsvd2:12; - - u32 params_fixup:8; - u32 converter:8; - u32 input_pin_type:1; - u32 output_pin_type:1; - u32 is_dynamic_in_pin:1; - u32 is_dynamic_out_pin:1; - u32 is_loadable:1; - u32 rsvd3:11; + __u16 module_id; + __u16 instance_id; + __u32 max_mcps; + __u32 mem_pages; + __u32 obs; + __u32 ibs; + __u32 vbus_id; + + __u32 max_in_queue:8; + __u32 max_out_queue:8; + __u32 time_slot:8; + __u32 core_id:4; + __u32 rsvd1:4; + + __u32 module_type:8; + __u32 conn_type:4; + __u32 dev_type:4; + __u32 hw_conn_type:4; + __u32 rsvd2:12; + + __u32 params_fixup:8; + __u32 converter:8; + __u32 input_pin_type:1; + __u32 output_pin_type:1; + __u32 is_dynamic_in_pin:1; + __u32 is_dynamic_out_pin:1; + __u32 is_loadable:1; + __u32 rsvd3:11; struct skl_dfw_v4_pipe pipe; struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE]; -- 2.18.0

1 0

[alsa-devel] [PATCH] fix ifnullfree.cocci warnings
by Julia Lawall 14 Aug '18

14 Aug '18

From: kbuild test robot <fengguang.wu(a)intel.com> NULL check before some freeing functions is not needed. Based on checkpatch warning "kfree(NULL) is safe this check is probably not required" and kfreeaddr.cocci by Julia Lawall. Generated by: scripts/coccinelle/free/ifnullfree.cocci Fixes: 0099cc17a399 ("ASoC:topology:avoid error log and oops during topology free.") Signed-off-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- Feel free to ignore this if the null test is useful in some way. topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2165,8 +2165,7 @@ void snd_sof_free_topology(struct snd_so /* free sroute and its private data */ kfree(sroute->route->source); kfree(sroute->route->sink); - if (sroute->route->control) - kfree(sroute->route->control); + kfree(sroute->route->control); kfree(sroute->route); kfree(sroute->private); kfree(sroute);

2 2

[alsa-devel] BUG of new AC97 bus layer
by Lihua Yao 14 Aug '18

14 Aug '18

Hi Robert; I am playing with AC97+WM9714 of TINY6410 board. I had made it to retrieve WM8914's vendor id but a BUG at probing WM97xx MFD device. After digging, I thought the offending code is 'structure assignment' at line 74. Both ac97->dev and adev->dev are objects of type 'struct device' rather than 'struct device *'. 65 struct snd_ac97 *snd_ac97_compat_alloc(struct ac97_codec_device *adev) 66 { 67 struct snd_ac97 *ac97; 68 69 ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); 70 if (ac97 == NULL) 71 return ERR_PTR(-ENOMEM); 72 73 printk("%s: %s %px\n", __func__, dev_name(&adev->dev), &adev->dev); 74 ac97->dev = adev->dev; 75 Below is a quick hack I came up with. diff --git a/sound/ac97/snd_ac97_compat.c b/sound/ac97/snd_ac97_compat.c index 61544e0d8de4..1ba2e7f48b41 100644 --- a/sound/ac97/snd_ac97_compat.c +++ b/sound/ac97/snd_ac97_compat.c @@ -70,7 +70,9 @@ struct snd_ac97 *snd_ac97_compat_alloc(struct ac97_codec_device *adev) if (ac97 == NULL) return ERR_PTR(-ENOMEM); - ac97->dev = adev->dev; + device_initialize(&ac97->dev); + dev_set_name(&ac97->dev, "%s-compat", dev_name(&adev->dev)); + ac97->private_data = adev; ac97->bus = &compat_soc_ac97_bus; return ac97; [ 2.657160] ac97_bus_scan: 574d4c13 [ 2.661390] ac97_bus_probe: 7f001000.ac97:0 [ 2.665256] snd_ac97_compat_alloc: 7f001000.ac97:0 cf0d3000 [ 2.670349] wm97xx-core 7f001000.ac97:0: wm97xx core found, id=0x574d4c13 [ 2.677338] __devm_regmap_init_ac97: 7f001000.ac97:0 cf2c4dd0 [ 2.682838] __devm_regmap_init: 7f001000.ac97:0 cf2c4dd0 [ 2.688795] list_add corruption. prev->next should be next (cf2c4f60), but was cf0d3190. (prev=cf20b480). [ 2.697775] ------------[ cut here ]------------ [ 2.702262] kernel BUG at /workspace/tiny6410/kernel/linux-stable/lib/list_debug.c:28! [ 2.710156] Internal error: Oops - BUG: 0 [#1] PREEMPT ARM [ 2.715622] Modules linked in: [ 2.718667] CPU: 0 PID: 1 Comm: swapper Not tainted 4.18.0-rc8 #100 [ 2.724904] Hardware name: Samsung S3C64xx (Flattened Device Tree) [ 2.731093] PC is at __list_add_valid+0x68/0x78 [ 2.735588] LR is at __list_add_valid+0x68/0x78 [ 2.740089] pc : [<c02ec4ac>] lr : [<c02ec4ac>] psr: 20000093 [ 2.746336] sp : cf887c18 ip : 60000093 fp : 00000000 [ 2.751541] r10: 00000000 r9 : cf2c4dd0 r8 : cf2b5e00 [ 2.756750] r7 : cf2c4f60 r6 : cf20b480 r5 : cf2c4dd0 r4 : cf1250c0 [ 2.763255] r3 : 00000001 r2 : cf886000 r1 : cf886004 r0 : 0000005d [ 2.769766] Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment none [ 2.776966] Control: 00c5387d Table: 50004008 DAC: 00000051 [ 2.782693] Process swapper (pid: 1, stack limit = 0x(ptrval)) [ 2.788505] Stack: (0xcf887c18 to 0xcf888000) [ 2.792848] 7c00: cf1250c0 c034dbf8 [ 2.801013] 7c20: a0000013 cf1250c0 a0000013 cf2c4f50 cf2c4dd0 a0000013 cf1250d0 c034e318 [ 2.809168] 7c40: cf2b5e00 cf2c4dd0 00000000 cf2c4c00 cf2b5e00 c03629e4 0000002f cf2b5e00 [ 2.817324] 7c60: c05d2ba8 c03634f4 00000000 00000000 00000000 c005b968 cf2b5e9c cf2b5e94 [ 2.825481] 7c80: c05d13c8 cf2c4dd0 cf125730 cf2c4c00 c05d13c8 c05d2ba8 00000000 00000000 [ 2.833637] 7ca0: 00000000 c03637fc 00000000 00000000 cf2c4c00 cf2c4dd0 00000000 c05d2ba8 [ 2.841793] 7cc0: 00000000 c03692b4 00000000 00000000 cf20b490 c0838700 cf20b49c c08aa1dc [ 2.849947] 7ce0: c083859c c036e8a4 00000001 00000001 00000000 c0581218 cf0d3000 c083859c [ 2.858105] 7d00: cf0d3104 c0428668 cf0d3000 00000000 c08aa1d4 c034ac64 cf0d3190 00000000 [ 2.866261] 7d20: cf887d58 c034ae18 00000000 00000000 cf15c010 00000000 c08aa1b0 c03491b0 [ 2.874418] 7d40: cf9e4a90 cf081ac4 cf0d3000 cf0d3034 00000001 c034a96c cf0d3000 00000001 [ 2.882574] 7d60: cf0d3000 cf0d3000 cf0d3000 c0840e40 00000000 c0349e80 cf0d3000 cf0d3008 [ 2.890729] 7d80: c07db048 c0348304 cf0d3008 cf15c000 cf125fc0 c0840e40 cf15c1e0 cf260a16 [ 2.898886] 7da0: cf15c000 cf15c000 cf15c000 00000000 c0840e40 cf15c1e0 c072d458 cf15c010 [ 2.907042] 7dc0: 000001d4 c0428b00 006000c0 cf91eb90 cf91ea10 c042882c 574d4c13 cf0d3000 [ 2.915198] 7de0: c08594c0 cf15c004 c05e6318 cf15ee10 cf91ea10 cf91ea00 00000000 00000000 [ 2.923354] 7e00: 00000000 cf15ee88 c08594c0 c0427a1c 00000000 cf9140a0 cf15ee10 00000000 [ 2.931510] 7e20: 00000000 cf91ea10 00000000 c0840a34 c0840a20 c0840a34 00000000 c034c8bc [ 2.939667] 7e40: c034c870 cf91ea10 00000000 c08aa1d4 c08aa1dc c034ac64 cf91eba0 00000000 [ 2.947822] 7e60: cf91ea10 c0840a34 cf91ea44 00000000 00000000 c07cc864 c0711df4 c034ade4 [ 2.955979] 7e80: c0840a34 c034acfc cf91ea10 c0349244 cf83ce70 cf8fa5e0 c0840a34 cf206360 [ 2.964133] 7ea0: c0837090 c034a0d4 c070505c c0840a34 00000006 c0840a34 00000006 c07db048 [ 2.972290] 7ec0: c0850ac0 c034b96c c034c20c c07c1e78 00000006 c000aa88 00000001 00000001 [ 2.980444] 7ee0: 00000000 00000000 00000000 00000000 c07e3d50 00000000 00000000 00000000 [ 2.988602] 7f00: cfffcda7 00000000 0000009f 0000009f cfffcdae c003bbcc c07600a0 00000000 [ 2.996757] 7f20: 00000006 00000006 cfffcdaa cfffcda7 00000000 c07cc838 00000006 cf260a16 [ 3.004915] 7f40: c07cc83c 00000006 c07d5744 c0850aec 00000000 c0790fec 00000006 00000006 [ 3.013069] 7f60: 00000000 c07905e0 00000000 0000009f 00000000 00000000 00000000 00000000 [ 3.021225] 7f80: 00000000 00000000 c057ae78 00000000 00000000 00000000 00000000 00000000 [ 3.029381] 7fa0: 00000000 c057ae80 00000000 c00090e8 00000000 00000000 00000000 00000000 [ 3.037536] 7fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 3.045693] 7fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 [ 3.053880] [<c02ec4ac>] (__list_add_valid) from [<c034dbf8>] (add_dr+0x38/0x58) [ 3.061232] [<c034dbf8>] (add_dr) from [<c034e318>] (devres_add+0x28/0x38) [ 3.068095] [<c034e318>] (devres_add) from [<c03629e4>] (regmap_attach_dev+0x44/0x64) [ 3.075900] [<c03629e4>] (regmap_attach_dev) from [<c03634f4>] (__regmap_init+0xa10/0xc94) [ 3.084139] [<c03634f4>] (__regmap_init) from [<c03637fc>] (__devm_regmap_init+0x84/0xc0) [ 3.092302] [<c03637fc>] (__devm_regmap_init) from [<c03692b4>] (__devm_regmap_init_ac97+0x54/0x68) [ 3.101325] [<c03692b4>] (__devm_regmap_init_ac97) from [<c036e8a4>] (wm97xx_ac97_probe+0xe0/0x170) [ 3.110351] [<c036e8a4>] (wm97xx_ac97_probe) from [<c0428668>] (ac97_bus_probe+0x8c/0x128) [ 3.118598] [<c0428668>] (ac97_bus_probe) from [<c034ac64>] (driver_probe_device+0x270/0x308) [ 3.127093] [<c034ac64>] (driver_probe_device) from [<c03491b0>] (bus_for_each_drv+0x5c/0x88) [ 3.135594] [<c03491b0>] (bus_for_each_drv) from [<c034a96c>] (__device_attach+0x9c/0xfc) [ 3.143748] [<c034a96c>] (__device_attach) from [<c0349e80>] (bus_probe_device+0x84/0x8c) [ 3.151904] [<c0349e80>] (bus_probe_device) from [<c0348304>] (device_add+0x32c/0x558) [ 3.159803] [<c0348304>] (device_add) from [<c0428b00>] (snd_ac97_controller_register+0x228/0x2a8) [ 3.168740] [<c0428b00>] (snd_ac97_controller_register) from [<c0427a1c>] (samsung_ac97_probe+0x19c/0x288) [ 3.178366] [<c0427a1c>] (samsung_ac97_probe) from [<c034c8bc>] (platform_drv_probe+0x4c/0x9c) [ 3.186959] [<c034c8bc>] (platform_drv_probe) from [<c034ac64>] (driver_probe_device+0x270/0x308) [ 3.195811] [<c034ac64>] (driver_probe_device) from [<c034ade4>] (__driver_attach+0xe8/0xec) [ 3.204224] [<c034ade4>] (__driver_attach) from [<c0349244>] (bus_for_each_dev+0x68/0x7c) [ 3.212380] [<c0349244>] (bus_for_each_dev) from [<c034a0d4>] (bus_add_driver+0x164/0x1f4) [ 3.220626] [<c034a0d4>] (bus_add_driver) from [<c034b96c>] (driver_register+0x7c/0x110) [ 3.228697] [<c034b96c>] (driver_register) from [<c000aa88>] (do_one_initcall+0x54/0x254) [ 3.236866] [<c000aa88>] (do_one_initcall) from [<c0790fec>] (kernel_init_freeable+0x260/0x340) [ 3.245540] [<c0790fec>] (kernel_init_freeable) from [<c057ae80>] (kernel_init+0x8/0xf0) [ 3.253598] [<c057ae80>] (kernel_init) from [<c00090e8>] (ret_from_fork+0x14/0x2c) [ 3.261125] Exception stack(0xcf887fb0 to 0xcf887ff8) [ 3.266164] 7fa0: 00000000 00000000 00000000 00000000 [ 3.274324] 7fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 3.282477] 7fe0: 00000000 00000000 00000000 00000000 00000013 00000000 [ 3.289079] Code: e1a01003 e59f0010 e1a0300c ebf5be50 (e7f001f2) [ 3.295163] ---[ end trace e28d0ebaa150deac ]--- Best Regard Lihua

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[alsa-devel] [PATCH - hwmixvolume 1/9] hwmixvolume: replace PyGTK with gobject-introspection
by Emmanuel Gil Peyrot 14 Aug '18

14 Aug '18

This doesn’t work yet, we require Gtk 3.0 rather than 2.0 and the API changed quite a lot. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index ef80bc8..28ce776 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -15,7 +15,10 @@ # OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR # PERFORMANCE OF THIS SOFTWARE. -import gobject, gtk +import gi +gi.require_version('GLib', '2.0') +gi.require_version('Gtk', '3.0') +from gi.repository import GLib, Gtk from pyalsa import alsacard, alsahcontrol INTF_PCM = alsahcontrol.interface_id['PCM'] @@ -57,16 +60,16 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) + self.label = Gtk.Label(self.get_label(info)) self.label.set_single_line_mode(True) self.parent.scales_vbox.pack_start(self.label, expand=False) for i in range(info.count): - adj = gtk.Adjustment(value=values[i], + adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) + scale = Gtk.HScale(adj) scale.set_draw_value(False) self.parent.scales_vbox.pack_start(scale, expand=False) self.scales.append(scale) @@ -157,7 +160,7 @@ class Stream: f.close() return cmdline.replace('\x00', ' ').strip() -class MixerWindow(gtk.Window): +class MixerWindow(Gtk.Window): card_numbers = alsacard.card_list() current_card = -1 hcontrol = None @@ -167,21 +170,21 @@ class MixerWindow(gtk.Window): hctl_sources = [] def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) + Gtk.Window.__init__(self) + self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = gtk.VBox() + vbox = Gtk.VBox() self.add(vbox) - hbox = gtk.HBox() + hbox = Gtk.HBox() vbox.pack_start(hbox, expand=False) - label = gtk.Label("_Sound Card: ") + label = Gtk.Label("_Sound Card: ") label.set_use_underline(True) hbox.pack_start(label, expand=False) - combo = gtk.combo_box_new_text() + combo = Gtk.combo_box_new_text() for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) @@ -191,23 +194,23 @@ class MixerWindow(gtk.Window): hbox.pack_start(combo) label.set_mnemonic_widget(combo) - self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton(label="_Lock Channels") self.lock_check.set_active(True) vbox.pack_start(self.lock_check, expand=False) - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) + scrollwin = Gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(Gtk.SHADOW_NONE) vbox.pack_start(scrollwin) - self.scales_vbox = gtk.VBox() + self.scales_vbox = Gtk.VBox() scrollwin.add_with_viewport(self.scales_vbox) - label = gtk.Label() + label = Gtk.Label() label.set_single_line_mode(True) line_height = label.size_request()[1] label.destroy() - scale = gtk.HScale() + scale = Gtk.HScale() scale.set_draw_value(False) line_height += scale.size_request()[1] scale.destroy() @@ -223,7 +226,7 @@ class MixerWindow(gtk.Window): def change_card(self, cardnum): for s in self.hctl_sources: - gobject.source_remove(s) + GLib.source_remove(s) self.hctl_sources = [] self.hcontrol = self.open_hcontrol_for_card(cardnum) @@ -249,7 +252,7 @@ class MixerWindow(gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) self.update_msg_label() @@ -267,7 +270,7 @@ class MixerWindow(gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = gtk.Label(msg) + self.msg_label = Gtk.Label(msg) self.scales_vbox.pack_start(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: @@ -280,9 +283,9 @@ class MixerWindow(gtk.Window): mode=alsahcontrol.open_mode['NONBLOCK']) except: # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + dlg = Gtk.MessageDialog(self, + Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, + Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() @@ -304,7 +307,7 @@ class MixerWindow(gtk.Window): def main(): MixerWindow() - gtk.main() + Gtk.main() main() -- 2.18.0

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