September 2018 - Alsa-devel - mailman.alsa-project.org

[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

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[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

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[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

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[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

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