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[alsa-devel] [PATCH] ASoC: max98357a: add missing supported rates
by Jerome Brunet 04 Apr '19

04 Apr '19

According the publicly available datasheet (and some test) the max98357a also supports 32, 44.1 and 88.2 kHz sample rate. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Datasheet is available here: https://datasheets.maximintegrated.com/en/ds/MAX98357A-MAX98357B.pdf sound/soc/codecs/max98357a.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d469576b5a7b..d037a3e4d323 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -97,7 +97,10 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 8000, .rate_max = 96000, -- 2.20.1

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[alsa-devel] [Xen-devel][PATCH] ALSA: xen-front: Do not use stream buffer size before it is set
by Oleksandr Andrushchenko 04 Apr '19

04 Apr '19

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> This fixes the regression introduced while moving to Xen shared buffer implementation. Fixes: 58f9d806d16a ("ALSA: xen-front: Use Xen common shared buffer implementation") Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> --- sound/xen/xen_snd_front_alsa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index a7f413cb704d..b14ab512c2ce 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -441,7 +441,7 @@ static int shbuf_setup_backstore(struct xen_snd_front_pcm_stream_info *stream, { int i; - stream->buffer = alloc_pages_exact(stream->buffer_sz, GFP_KERNEL); + stream->buffer = alloc_pages_exact(buffer_sz, GFP_KERNEL); if (!stream->buffer) return -ENOMEM; -- 2.21.0

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[alsa-devel] [PATCH] ASoC: imx-pcm: Switch to SPDX identifier
by Andra Danciu 04 Apr '19

04 Apr '19

Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu <andradanciu1997(a)gmail.com> --- sound/soc/fsl/imx-pcm.h | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 133c4470acad..3ce2f437e645 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -1,13 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> * * This code is based on code copyrighted by Freescale, * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. */ #ifndef _IMX_PCM_H -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: imx-pcm-dma: Switch to SPDX identifier
by Andra Danciu 04 Apr '19

04 Apr '19

Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Andra Danciu <andradanciu1997(a)gmail.com> --- sound/soc/fsl/imx-pcm-dma.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 314814ddd2b0..6c495761f33c 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -1,16 +1,12 @@ -/* - * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer - * - * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> - * - * This code is based on code copyrighted by Freescale, - * Liam Girdwood, Javier Martin and probably others. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer +// +// Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> +// +// This code is based on code copyrighted by Freescale, +// Liam Girdwood, Javier Martin and probably others. + #include <linux/platform_device.h> #include <linux/dmaengine.h> #include <linux/types.h> -- 2.11.0

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[alsa-devel] [PATCH 4.14 085/121] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

4.14-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2db4d0c80d33..393100edd5fd 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH 5.0 173/246] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

5.0-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 81f2fe2c6d23..60f87a0d99f4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

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[alsa-devel] [PATCH 4.9 66/91] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

4.9-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index dffd549a0e2a..705d2524ec31 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH 4.19 131/187] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

4.19-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 44433b20435c..600d9be9706e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH 4.9 44/91] SoC: imx-sgtl5000: add missing put_device()
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

4.9-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..8e525f7ac08d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,6 +115,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH 4.14 058/121] SoC: imx-sgtl5000: add missing put_device()
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

4.14-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..8e525f7ac08d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,6 +115,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH 4.19 083/187] SoC: imx-sgtl5000: add missing put_device()
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

4.19-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..9b9a7ec52905 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -108,6 +108,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

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[alsa-devel] [PATCH 5.0 111/246] SoC: imx-sgtl5000: add missing put_device()
by Greg Kroah-Hartman 04 Apr '19

04 Apr '19

5.0-stable review patch. If anyone has any objections, please let me know. ------------------ [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..9b9a7ec52905 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -108,6 +108,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

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[alsa-devel] Applied "ASoC: audio-graph-card: don't select DPCM via audio-graph-card" to the asoc tree
by Mark Brown 04 Apr '19

04 Apr '19

The patch ASoC: audio-graph-card: don't select DPCM via audio-graph-card 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 c2c616021d64d952dc9d37793924ce57833d7754 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 4 Apr 2019 09:52:15 +0900 Subject: [PATCH] ASoC: audio-graph-card: don't select DPCM via audio-graph-card commit ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") merged audio-graph-scu-card which can handle DPCM into audio-graph-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its OF-graph endpoint connection. But, because of it, existing "audio-graph-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "audio-graph-card" user can select "normal sound card", and "audio-graph-scu-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") Reported-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351330e8..69bc4848d787 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,6 +20,8 @@ #include <linux/string.h> #include <sound/simple_card_utils.h> +#define DPCM_SELECTABLE 1 + struct graph_priv { struct snd_soc_card snd_card; struct graph_dai_props { @@ -440,6 +442,7 @@ static int graph_for_each_link(struct graph_priv *priv, struct device_node *codec_port; struct device_node *codec_port_old = NULL; struct asoc_simple_card_data adata; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); int rc, ret; /* loop for all listed CPU port */ @@ -470,8 +473,9 @@ static int graph_for_each_link(struct graph_priv *priv, * if Codec port has many endpoints, * or has convert-xxx property */ - if ((of_get_child_count(codec_port) > 1) || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, cpu_ep, codec_ep, li, (codec_port_old == codec_port)); /* else normal sound */ @@ -732,7 +736,8 @@ static int graph_remove(struct platform_device *pdev) static const struct of_device_id graph_of_match[] = { { .compatible = "audio-graph-card", }, - { .compatible = "audio-graph-scu-card", }, + { .compatible = "audio-graph-scu-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, graph_of_match); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card: don't select DPCM via simple-audio-card" to the asoc tree
by Mark Brown 04 Apr '19

04 Apr '19

The patch ASoC: simple-card: don't select DPCM via simple-audio-card 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 42bf029a55a9bb8036f1d738a28dba2f7ec1e79d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 4 Apr 2019 09:52:52 +0900 Subject: [PATCH] ASoC: simple-card: don't select DPCM via simple-audio-card commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-scu-audio-card which can handle DPCM into simple-audio-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its CPU/Codec DAI count. But, because of it, existing "simple-audio-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "simple-audio-card" user can select "normal sound card", and "simple-scu-audio-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: da215354eb55c ("ASoC: simple-card: merge simple-scu-card") 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.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7147bba45a2a..34de32efc4c4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,12 +9,15 @@ #include <linux/device.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/string.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> +#define DPCM_SELECTABLE 1 + struct simple_priv { struct snd_soc_card snd_card; struct simple_dai_props { @@ -441,6 +444,7 @@ static int simple_for_each_link(struct simple_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; struct device_node *node; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); bool is_top = 0; int ret = 0; @@ -480,8 +484,9 @@ static int simple_for_each_link(struct simple_priv *priv, * if it has many CPUs, * or has convert-xxx property */ - if (num > 2 || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + (num > 2 || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, np, codec, li, is_top); /* else normal sound */ else @@ -822,7 +827,8 @@ static int simple_remove(struct platform_device *pdev) static const struct of_device_id simple_of_match[] = { { .compatible = "simple-audio-card", }, - { .compatible = "simple-scu-audio-card", }, + { .compatible = "simple-scu-audio-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, simple_of_match); -- 2.20.1

1 0

[alsa-devel] [PATCH v1 0/12] Patches to update pdm for rockchip socs
by Sugar Zhang 04 Apr '19

04 Apr '19

These patches to add support pdm for rk322x/rk1808/rk3308/px30 socs and fix some issues. Sugar Zhang (12): ASoC: rockchip: pdm: fix regmap_ops hang issue ASoC: rockchip: pdm: using left justified store mode ASoC: rockchip: pdm: add default regs ASoC: rockchip: pdm: optimize clear logic ASoC: rockchip: pdm: change dma burst to 8 ASoC: rockchip: pdm: fixup pdm fractional div dt-bindings: sound: rockchip: add compatible for rk3308/px30 ASoC: rockchip: pdm: add compatible for rk1808 dt-bindings: sound: add compatible for rk1808 ASoC: rockchip: pdm: adjust waterlevel in frame unit ASoC: rockchip: pdm: Mark RXFIFO_DATA as volatile and precious ASoC: rockchip: pdm: Correct PDM_CTRL0 reg value .../devicetree/bindings/sound/rockchip,pdm.txt | 5 + sound/soc/rockchip/rockchip_pdm.c | 209 ++++++++++++++++----- sound/soc/rockchip/rockchip_pdm.h | 12 ++ 3 files changed, 180 insertions(+), 46 deletions(-) -- 2.7.4

2 25

[alsa-devel] [PATCH][next] ASoC: sprd: Fix spelling mistake "faied" -> "failed"
by Colin King 04 Apr '19

04 Apr '19

From: Colin Ian King <colin.king(a)canonical.com> There are two identical spelling mistakes in dev_err messages. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sprd/sprd-pcm-compress.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index dc84257cffc2..e008525a6cb3 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -282,13 +282,13 @@ static int sprd_platform_compr_set_params(struct snd_compr_stream *cstream, */ ret = sprd_platform_compr_dma_config(cstream, params, 1); if (ret) { - dev_err(dev, "faied to config stage 1 DMA: %d\n", ret); + dev_err(dev, "failed to config stage 1 DMA: %d\n", ret); return ret; } ret = sprd_platform_compr_dma_config(cstream, params, 0); if (ret) { - dev_err(dev, "faied to config stage 0 DMA: %d\n", ret); + dev_err(dev, "failed to config stage 0 DMA: %d\n", ret); goto config_err; } -- 2.20.1

4 3

[alsa-devel] [PATCH V2] ASoC: fsl_esai: Support synchronous mode
by S.j. Wang 04 Apr '19

04 Apr '19

In ESAI synchronous mode, the clock is generated by Tx, So we should always set registers of Tx which relate with the bit clock and frame clock generation (TCCR, TCR, ECR), even there is only Rx is working. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - refine the patch according Nicolin's comments - merge plain setting. sound/soc/fsl/fsl_esai.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..f2e7b27f8447 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -218,7 +218,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); struct clk *clksrc = esai_priv->extalclk; - bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool tx = (clk_id <= ESAI_HCKT_EXTAL || esai_priv->synchronous); bool in = dir == SND_SOC_CLOCK_IN; u32 ratio, ecr = 0; unsigned long clk_rate; @@ -253,7 +253,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, ecr |= ESAI_ECR_ETI; /* fall through */ case ESAI_HCKR_EXTAL: - ecr |= ESAI_ECR_ERI; + ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; default: return -EINVAL; @@ -537,10 +537,18 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bclk = params_rate(params) * slot_width * esai_priv->slots; - ret = fsl_esai_set_bclk(dai, tx, bclk); + ret = fsl_esai_set_bclk(dai, esai_priv->synchronous || tx, bclk); if (ret) return ret; + mask = ESAI_xCR_xSWS_MASK; + val = ESAI_xCR_xSWS(slot_width, width); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + /* Recording in synchronous mode needs to set TCR also */ + if (!tx && esai_priv->synchronous) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, val); + /* Use Normal mode to support monaural audio */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? @@ -556,10 +564,9 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); - mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); - val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0); - - regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + if (tx) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_PADC, ESAI_xCR_PADC); /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */ regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, -- 1.9.1

2 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic32x4: Change author's name
by Annaliese McDermond 04 Apr '19

04 Apr '19

The author of these files has changed her name. Update instances in the code of her dead name to current legal name. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4-i2c.c | 4 ++-- sound/soc/codecs/tlv320aic32x4-spi.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c index 385fa2e9525a..22c3a6bc0b6c 100644 --- a/sound/soc/codecs/tlv320aic32x4-i2c.c +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -3,7 +3,7 @@ * * Copyright 2011 NW Digital Radio * - * Author: Jeremy McDermond <nh6z(a)nh6z.net> + * Author: Annaliese McDermond <nh6z(a)nh6z.net> * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * @@ -72,5 +72,5 @@ static struct i2c_driver aic32x4_i2c_driver = { module_i2c_driver(aic32x4_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C"); -MODULE_AUTHOR("Jeremy McDermond <nh6z(a)nh6z.net>"); +MODULE_AUTHOR("Annaliese McDermond <nh6z(a)nh6z.net>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index 07d78ae51e05..aa5b7ba0254b 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -3,7 +3,7 @@ * * Copyright 2011 NW Digital Radio * - * Author: Jeremy McDermond <nh6z(a)nh6z.net> + * Author: Annaliese McDermond <nh6z(a)nh6z.net> * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * @@ -74,5 +74,5 @@ static struct spi_driver aic32x4_spi_driver = { module_spi_driver(aic32x4_spi_driver); MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI"); -MODULE_AUTHOR("Jeremy McDermond <nh6z(a)nh6z.net>"); +MODULE_AUTHOR("Annaliese McDermond <nh6z(a)nh6z.net>"); MODULE_LICENSE("GPL"); -- 2.19.1

2 2

[alsa-devel] Applied "ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings" to the asoc tree
by Mark Brown 04 Apr '19

04 Apr '19

The patch ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings 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 b6c573e5e8652330525d61b583203e7b71a09c53 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Wed, 3 Apr 2019 07:11:39 +0000 Subject: [PATCH] ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index a50c6af5c945..681bedb524f9 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -364,7 +364,6 @@ static const struct of_device_id mt8183_mt6358_ts3a227_max98357_dt_match[] = { static struct platform_driver mt8183_mt6358_ts3a227_max98357_driver = { .driver = { .name = "mt8183_mt6358_ts3a227_max98357", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt8183_mt6358_ts3a227_max98357_dt_match, #endif -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings" to the asoc tree
by Mark Brown 04 Apr '19

04 Apr '19

The patch ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings 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 d82e78b0a578a712629cdb6cb7bfc66fb3215306 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Wed, 3 Apr 2019 07:11:59 +0000 Subject: [PATCH] ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 1e7e8aed8707..31ea8632c397 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -454,7 +454,6 @@ static const struct of_device_id mt8183_da7219_max98357_dt_match[] = { static struct platform_driver mt8183_da7219_max98357_driver = { .driver = { .name = "mt8183_da7219_max98357", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt8183_da7219_max98357_dt_match, #endif -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: sprd: Fix the smatch warning
by Baolin Wang 04 Apr '19

04 Apr '19

Remove the unnecessary validation of the 'cstream' variable to fix below smatch warning: sprd_platform_compr_drain_notify() warn: variable dereferenced before check 'cstream' (see line 105) Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> --- sound/soc/sprd/sprd-pcm-compress.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index dc84257..424e983 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -107,8 +107,7 @@ static void sprd_platform_compr_drain_notify(void *arg) memset(stream->info_area, 0, sizeof(struct sprd_compr_playinfo)); - if (cstream) - snd_compr_drain_notify(cstream); + snd_compr_drain_notify(cstream); } static void sprd_platform_compr_dma_complete(void *data) -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: tlv320aic32x4: Update copyright and use SPDX identifier" to the asoc tree
by Mark Brown 04 Apr '19

04 Apr '19

The patch ASoC: tlv320aic32x4: Update copyright and use SPDX identifier 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 8a1d95c393d971e624fc28f11516b0bc3a7fa706 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Wed, 3 Apr 2019 21:17:16 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Update copyright and use SPDX identifier Update the copyright dates and use the SPDX identifier instead of reciting the license. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-i2c.c | 14 ++------------ sound/soc/codecs/tlv320aic32x4-spi.c | 14 ++------------ 2 files changed, 4 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c index 385fa2e9525a..d8bede5c1f38 100644 --- a/sound/soc/codecs/tlv320aic32x4-i2c.c +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -1,21 +1,11 @@ -/* - * linux/sound/soc/codecs/tlv320aic32x4-i2c.c +/* SPDX-License-Identifier: GPL-2.0 * - * Copyright 2011 NW Digital Radio + * Copyright 2011-2019 NW Digital Radio * * Author: Jeremy McDermond <nh6z(a)nh6z.net> * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. */ #include <linux/i2c.h> diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index 07d78ae51e05..a72a61c4d785 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -1,21 +1,11 @@ -/* - * linux/sound/soc/codecs/tlv320aic32x4-spi.c +/* SPDX-License-Identifier: GPL-2.0 * - * Copyright 2011 NW Digital Radio + * Copyright 2011-2019 NW Digital Radio * * Author: Jeremy McDermond <nh6z(a)nh6z.net> * * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. */ #include <linux/spi/spi.h> -- 2.20.1

1 0

[alsa-devel] [PATCH -next] ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings
by YueHaibing 04 Apr '19

04 Apr '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 93536659a859..117d81b43542 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -454,7 +454,6 @@ static const struct of_device_id mt8183_da7219_max98357_dt_match[] = { static struct platform_driver mt8183_da7219_max98357_driver = { .driver = { .name = "mt8183_da7219_max98357", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt8183_da7219_max98357_dt_match, #endif

1 2

[alsa-devel] [PATCH -next] ASoC: Mediatek: MT8183: Fix platform_no_drv_owner.cocci warnings
by YueHaibing 04 Apr '19

04 Apr '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 2da56232a9e1..7ed3c947276d 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -364,7 +364,6 @@ static const struct of_device_id mt8183_mt6358_ts3a227_max98357_dt_match[] = { static struct platform_driver mt8183_mt6358_ts3a227_max98357_driver = { .driver = { .name = "mt8183_mt6358_ts3a227_max98357", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt8183_mt6358_ts3a227_max98357_dt_match, #endif

1 2

[alsa-devel] [PATCH v2] ASoC: simple-card: don't select DPCM via simple-audio-card
by Kuninori Morimoto 04 Apr '19

04 Apr '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-scu-audio-card which can handle DPCM into simple-audio-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its CPU/Codec DAI count. But, because of it, existing "simple-audio-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "simple-audio-card" user can select "normal sound card", and "simple-scu-audio-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: da215354eb55c ("ASoC: simple-card: merge simple-scu-card") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is for v5.1 v1 -> v2 - add Fixes: tag sound/soc/generic/simple-card.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7147bba..34de32e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,12 +9,15 @@ #include <linux/device.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/string.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> +#define DPCM_SELECTABLE 1 + struct simple_priv { struct snd_soc_card snd_card; struct simple_dai_props { @@ -441,6 +444,7 @@ static int simple_for_each_link(struct simple_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; struct device_node *node; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); bool is_top = 0; int ret = 0; @@ -480,8 +484,9 @@ static int simple_for_each_link(struct simple_priv *priv, * if it has many CPUs, * or has convert-xxx property */ - if (num > 2 || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + (num > 2 || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, np, codec, li, is_top); /* else normal sound */ else @@ -822,7 +827,8 @@ static int simple_remove(struct platform_device *pdev) static const struct of_device_id simple_of_match[] = { { .compatible = "simple-audio-card", }, - { .compatible = "simple-scu-audio-card", }, + { .compatible = "simple-scu-audio-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, simple_of_match); -- 2.7.4

1 0

[alsa-devel] [PATCH v2] ASoC: audio-graph-card: don't select DPCM via audio-graph-card
by Kuninori Morimoto 04 Apr '19

04 Apr '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") merged audio-graph-scu-card which can handle DPCM into audio-graph-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its OF-graph endpoint connection. But, because of it, existing "audio-graph-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "audio-graph-card" user can select "normal sound card", and "audio-graph-scu-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Fixes: ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") Reported-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Acked-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- This is for v5.1 v1 -> v2 - add Fixes: tag - add Arnaud's Acked-by sound/soc/generic/audio-graph-card.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351..69bc484 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,6 +20,8 @@ #include <linux/string.h> #include <sound/simple_card_utils.h> +#define DPCM_SELECTABLE 1 + struct graph_priv { struct snd_soc_card snd_card; struct graph_dai_props { @@ -440,6 +442,7 @@ static int graph_for_each_link(struct graph_priv *priv, struct device_node *codec_port; struct device_node *codec_port_old = NULL; struct asoc_simple_card_data adata; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); int rc, ret; /* loop for all listed CPU port */ @@ -470,8 +473,9 @@ static int graph_for_each_link(struct graph_priv *priv, * if Codec port has many endpoints, * or has convert-xxx property */ - if ((of_get_child_count(codec_port) > 1) || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, cpu_ep, codec_ep, li, (codec_port_old == codec_port)); /* else normal sound */ @@ -732,7 +736,8 @@ static int graph_remove(struct platform_device *pdev) static const struct of_device_id graph_of_match[] = { { .compatible = "audio-graph-card", }, - { .compatible = "audio-graph-scu-card", }, + { .compatible = "audio-graph-scu-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, graph_of_match); -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: audio-graph-card: don't select DPCM via audio-graph-card
by Kuninori Morimoto 04 Apr '19

04 Apr '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit ae3cb5790906b ("ASoC: audio-graph-card: merge audio-graph-scu-card") merged audio-graph-scu-card which can handle DPCM into audio-graph-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its OF-graph endpoint connection. But, because of it, existing "audio-graph-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "audio-graph-card" user can select "normal sound card", and "audio-graph-scu-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Reported-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351..69bc484 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,6 +20,8 @@ #include <linux/string.h> #include <sound/simple_card_utils.h> +#define DPCM_SELECTABLE 1 + struct graph_priv { struct snd_soc_card snd_card; struct graph_dai_props { @@ -440,6 +442,7 @@ static int graph_for_each_link(struct graph_priv *priv, struct device_node *codec_port; struct device_node *codec_port_old = NULL; struct asoc_simple_card_data adata; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); int rc, ret; /* loop for all listed CPU port */ @@ -470,8 +473,9 @@ static int graph_for_each_link(struct graph_priv *priv, * if Codec port has many endpoints, * or has convert-xxx property */ - if ((of_get_child_count(codec_port) > 1) || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + ((of_get_child_count(codec_port) > 1) || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, cpu_ep, codec_ep, li, (codec_port_old == codec_port)); /* else normal sound */ @@ -732,7 +736,8 @@ static int graph_remove(struct platform_device *pdev) static const struct of_device_id graph_of_match[] = { { .compatible = "audio-graph-card", }, - { .compatible = "audio-graph-scu-card", }, + { .compatible = "audio-graph-scu-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, graph_of_match); -- 2.7.4

3 4

[alsa-devel] [PATCH] ASoC: simple-card: don't select DPCM via simple-audio-card
by Kuninori Morimoto 04 Apr '19

04 Apr '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-scu-audio-card which can handle DPCM into simple-audio-card. By this patch, the judgement to select "normal sound card" or "DPCM sound card" is based on its CPU/Codec DAI count. But, because of it, existing "simple-audio-card" user who is assuming "normal sound card" might select DPCM unintentionally. To solve this issue, this patch allows "simple-audio-card" user can select "normal sound card", and "simple-scu-audio-card" user can select both "normal sound card" and "DPCM sound card". This keeps compatibility collectry. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7147bba..34de32e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,12 +9,15 @@ #include <linux/device.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/string.h> #include <sound/simple_card.h> #include <sound/soc-dai.h> #include <sound/soc.h> +#define DPCM_SELECTABLE 1 + struct simple_priv { struct snd_soc_card snd_card; struct simple_dai_props { @@ -441,6 +444,7 @@ static int simple_for_each_link(struct simple_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; struct device_node *node; + uintptr_t dpcm_selectable = (uintptr_t)of_device_get_match_data(dev); bool is_top = 0; int ret = 0; @@ -480,8 +484,9 @@ static int simple_for_each_link(struct simple_priv *priv, * if it has many CPUs, * or has convert-xxx property */ - if (num > 2 || - adata.convert_rate || adata.convert_channels) + if (dpcm_selectable && + (num > 2 || + adata.convert_rate || adata.convert_channels)) ret = func_dpcm(priv, np, codec, li, is_top); /* else normal sound */ else @@ -822,7 +827,8 @@ static int simple_remove(struct platform_device *pdev) static const struct of_device_id simple_of_match[] = { { .compatible = "simple-audio-card", }, - { .compatible = "simple-scu-audio-card", }, + { .compatible = "simple-scu-audio-card", + .data = (void *)DPCM_SELECTABLE }, {}, }; MODULE_DEVICE_TABLE(of, simple_of_match); -- 2.7.4

1 1

[alsa-devel] [GIT PULL FOR v5.2] Media Device Allocator API
by Hans Verkuil 04 Apr '19

04 Apr '19

Hi Mauro, After five (!) years of work, this is finally ready to go in. Thank you Shuah for your perseverance! Regards, Hans The following changes since commit 1c3ec30bb23023d738b538e64ac3028902d53692: media: index.rst: exclude Indexes section from latex/pdf output (2019-03-30 09:42:35 -0400) are available in the Git repository at: git://linuxtv.org/hverkuil/media_tree.git tags/br-v5.2c3 for you to fetch changes up to 5ea25de43fd59a319136cf5b2e547ca315cfb212: au0828: stop video streaming only when last user stops (2019-04-03 09:01:15 +0200) ---------------------------------------------------------------- Tag branch ---------------------------------------------------------------- Hans Verkuil (1): au0828: stop video streaming only when last user stops Shuah Khan (8): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies selftests: media_dev_allocator api test au0828: Fix NULL pointer dereference in au0828_analog_stream_enable() media: Clear devnode->media_dev holding media_devnode_lock Documentation/media/kapi/mc-core.rst | 41 ++++++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 135 +++++++++++++++++ drivers/media/media-devnode.c | 3 +- drivers/media/usb/au0828/Kconfig | 2 + drivers/media/usb/au0828/au0828-core.c | 196 ++++++++++++++++++------ drivers/media/usb/au0828/au0828-video.c | 16 +- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 63 ++++++++ include/uapi/linux/media.h | 25 ++-- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 327 +++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 +++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + tools/testing/selftests/media_tests/media_dev_allocator.sh | 85 +++++++++++ 22 files changed, 976 insertions(+), 67 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h create mode 100755 tools/testing/selftests/media_tests/media_dev_allocator.sh

2 1

[alsa-devel] [PATCH RFC] ALSA: usb-audio: Scarlett mixer interface for 18i20 Gen 2
by Geoffrey D. Bennett 03 Apr '19

03 Apr '19

Add mixer quirk for the Focusrite Scarlett 18i20 Gen 2 audio interface. Although the interface is USB compliant, additional hardware mixing, routing, and metering functionality is available using proprietary USB requests. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- Hi all, This patch adds the following controls for the Scarlett 18i20 Gen 2: - Master volume knob indicator - Volume controls for the 10 analogue HW outputs - HW/SW volume switches for the 10 analogue HW outputs - Output mux (where the sound for the HW outputs comes from; defaults to PCM outputs 1-20) - Capture mux (where the sound for PCM recording comes from; defaults to HW inputs 1-18) - Matrix mux (where the sound going into the mixer comes from; 18 inputs default off) - Mixer matrix (18 inputs * 10 outputs = 180 controls) - Mute and dim indicators - Level meters This is my first attempt at writing any kernel device driver code, so apologies in advance if I missed something. There's a couple of things I'm particularly unsure about: - I added a private field to struct snd_usb_audio for storing the private mixer data. This seems wrong, but I didn't know where else I could/should put that data. - I've #define'd the vendor-specific interface, endpoint, max_data, and interval values. Is it okay to leave these hard-coded (they match the values provided by lsusb -v; I presume they never change?), or should they be looked up at runtime instead? (and if at runtime, how?) - I'm not sure if those snd_printk()s for unexpected responses from the hardware should be left in there? Removed? Reworded? Comments and feedback would be very welcome. Regards, Geoffrey. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index d330f74c90e6..8cf7081e17cb 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -11,6 +11,7 @@ snd-usb-audio-objs := card.o \ mixer.o \ mixer_quirks.o \ mixer_scarlett.o \ + mixer_scarlett_gen2.o \ mixer_us16x08.o \ pcm.o \ power.o \ diff --git a/sound/usb/card.c b/sound/usb/card.c index 746a72e23cf9..06bd839e9ab2 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -354,6 +354,8 @@ static void snd_usb_audio_free(struct snd_card *card) list_for_each_entry_safe(ep, n, &chip->ep_list, list) snd_usb_endpoint_free(ep); + kfree(chip->private); + mutex_destroy(&chip->mutex); if (!atomic_read(&chip->shutdown)) dev_set_drvdata(&chip->dev->dev, NULL); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 85ae0ff2382a..331f0596e42d 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -45,6 +45,7 @@ #include "mixer.h" #include "mixer_quirks.h" #include "mixer_scarlett.h" +#include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" #include "helper.h" @@ -2273,6 +2274,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_scarlett_controls_create(mixer); break; + case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ + err = snd_scarlett_gen2_controls_create(mixer); + break; + case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ err = snd_soundblaster_e1_switch_create(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c new file mode 100644 index 000000000000..943b65ff27f1 --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.c @@ -0,0 +1,1592 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Focusrite Scarlett 18i20 Gen 2 Driver for ALSA + * + * Copyright (c) 2018-2019 by Geoffrey D. Bennett + * + * Based on the Scarlett (Gen 1) Driver for ALSA: + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus <robin at gareus.org> + * Copyright (c) 2002 by Takashi Iwai <tiwai at suse.de> + * Copyright (c) 2014 by Chris J Arges <chris.j.arges at canonical.com> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson <david.henningsson at canonical.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +/* Mixer Interface for the Focusrite Scarlett 18i20 Gen 2 audio + * interface. Based on the Gen 1 driver and rewritten. + */ + +/* The protocol was reverse engineered by looking at the communication + * between Focusrite Control 2.3.4 and the Focusrite(R) Scarlett 18i20 + * (firmware 1083) using usbmon in July-August 2018. + * + * This ALSA mixer gives access to: + * - input, output, mixer-matrix muxes + * - 18x10 mixer-matrix gain stages + * - gain/volume controls + * - level meters + * + * <ditaa> + * /--------------\ 18chn 20chn /--------------\ + * | Hardware in +--+------\ /-------------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | /-----\ | + * | | | | | | + * | v v v | | + * | +---------------+ | | + * | \ Matrix Mux / | | + * | +-----+-----+ | | + * | | | | + * | |18chn | | + * | | | | + * | | 10chn| | + * | v | | + * | +------------+ | | + * | | Mixer | | | + * | | Matrix | | | + * | | | | | + * | | 18x10 Gain | | | + * | | stages | | | + * | +-----+------+ | | + * | | | | + * |18chn |10chn | |20chn + * | | | | + * | +----------/ | + * | | | + * v v v + * =========================== + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +---+---+---+ +-----+-----+ + * | | | + * 10chn| | |18chn + * | | | + * /--------------\ | | | /--------------\ + * | S/PDIF, ADAT |<--/ |10chn \-->| ALSA PCM in | + * | Hardware out | | \--------------/ + * \--------------/ | + * v + * +-------------+ switch per channel to select + * | Master Gain |<-- front panel volume knob or + * +------+------+ individual software gain + * | + * |10chn + * /--------------\ | + * | Analogue |<------/ + * | Hardware out | + * \--------------/ + * </ditaa> + * + */ + +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/control.h> +#include <sound/tlv.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" + +#include "mixer_scarlett_gen2.h" + +/* some gui mixers can't handle negative ctl values */ +#define SCARLETT_VOLUME_BIAS 127 + +/* mixer range from -80dB to +6dB in 0.5dB steps */ +#define SCARLETT_MIXER_MIN_DB -80 +#define SCARLETT_MIXER_BIAS (-SCARLETT_MIXER_MIN_DB * 2) +#define SCARLETT_MIXER_MAX_DB 6 +#define SCARLETT_MIXER_MAX_VALUE \ + ((SCARLETT_MIXER_MAX_DB - SCARLETT_MIXER_MIN_DB) * 2) + +/* map from (dB + 80) * 2 to mixer value + * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + */ +static const __u16 scarlett2_mixer_values[173] = { + 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, + 2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 5, 5, 5, 6, 6, 6, 7, 7, 8, 8, + 9, 9, 10, 10, 11, 12, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, + 23, 24, 25, 27, 29, 30, 32, 34, 36, 38, 41, 43, 46, 48, 51, + 54, 57, 61, 65, 68, 73, 77, 81, 86, 91, 97, 103, 109, 115, + 122, 129, 137, 145, 154, 163, 173, 183, 194, 205, 217, 230, + 244, 259, 274, 290, 307, 326, 345, 365, 387, 410, 434, 460, + 487, 516, 547, 579, 614, 650, 689, 730, 773, 819, 867, 919, + 973, 1031, 1092, 1157, 1225, 1298, 1375, 1456, 1543, 1634, + 1731, 1833, 1942, 2057, 2179, 2308, 2445, 2590, 2744, 2906, + 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, + 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, + 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, + 16345 +}; + +/* Maximum number of analogue outputs */ +#define SCARLETT2_CONTROLS_MAX 10 + +/* Maximum number of inputs to the mixer */ +#define SCARLETT2_INPUT_MIX_MAX 18 + +/* Maximum number of outputs from the mixer */ +#define SCARLETT2_OUTPUT_MIX_MAX 10 + +/* Maximum size of the data in the USB mux assignment message: + * 18 inputs, 20 outputs, 18 matrix inputs, 8 spare + */ +#define SCARLETT2_MUX_MAX 64 + +/* Number of meters: + * 18 inputs, 20 outputs, 18 matrix inputs + */ +#define SCARLETT2_NUM_METERS 56 + +/* Hardware port types: + * - None (no input to mux) + * - Analogue I/O + * - S/PDIF I/O + * - ADAT I/O + * - Mixer I/O + * - PCM I/O + */ +enum { + SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_ANALOGUE = 1, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, +}; + +/* Count of total I/O and number available at each sample rate */ +enum { + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_OUT_176 = 4, + SCARLETT2_PORT_DIRECTIONS = 5, +}; + +#define SCARLETT2_BUTTON_COUNT 2 + +static const char *const scarlett2_button_names[SCARLETT2_BUTTON_COUNT] = { + "Mute", "Dim" +}; + +/* Description of each hardware port type: + * - id: hardware ID for this port type + * - num: number of sources/destinations of this port type + * - src_descr: printf format string for mux input selections + * - src_num_offset: added to channel number for the fprintf + * - dst_descr: printf format string for mixer controls + */ +struct scarlett2_ports { + __u16 id; + int num[SCARLETT2_PORT_DIRECTIONS]; + const char * const src_descr; + int src_num_offset; + const char * const dst_descr; +}; + +struct scarlett2_device_info { + const char * const line_out_descrs[SCARLETT2_CONTROLS_MAX]; + struct scarlett2_ports ports[SCARLETT2_PORT_TYPE_COUNT]; +}; + +struct scarlett2_chip_data { + struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct mutex data_mutex; /* lock access to this data */ + const struct scarlett2_device_info *info; + int num_mux_srcs; + __u16 scarlett2_seq; + __u8 vol_updated; + __u8 master_vol; + __u8 vol[SCARLETT2_CONTROLS_MAX]; + __u8 vol_sw_hw_switch[SCARLETT2_CONTROLS_MAX]; + __u8 buttons[SCARLETT2_BUTTON_COUNT]; + struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *vol_ctls[SCARLETT2_CONTROLS_MAX]; + struct snd_kcontrol *button_ctls[SCARLETT2_CONTROLS_MAX]; + __u8 mux[SCARLETT2_MUX_MAX]; + __u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; +}; + +struct __attribute__((__packed__)) scarlett2_usb_volume_status { + __u8 buttons[SCARLETT2_BUTTON_COUNT]; /* mute & dim buttons */ + __u8 pad1; + __s16 sw_vol[10]; /* software volume setting */ + __s16 hw_vol[10]; /* actual volume of output inc. dim (-18dB) */ + __u16 pad2[5]; + __u8 sw_hw_switch[10]; /* sw (0) or hw (1) controlled */ + __u16 pad3[3]; + __s16 master_vol; /* front panel volume knob */ +}; + +/*** Model-specific data ***/ + +static const struct scarlett2_device_info s18i20_gen2_info = { + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .ports = { + { + .id = 0x000, + .num = { 1, 0, 8, 8, 6 }, + .src_descr = "Off", + .src_num_offset = 0, + }, + { + .id = 0x080, + .num = { 8, 10, 10, 10, 10 }, + .src_descr = "Analogue %d", + .src_num_offset = 1, + .dst_descr = "Analogue Output %02d Playback" + }, + { + /* S/PDIF outputs aren't available at 192KHz + * but are included in the USB mux I/O + * assignment message anyway + */ + .id = 0x180, + .num = { 2, 2, 2, 2, 2 }, + .src_descr = "S/PDIF %d", + .src_num_offset = 1, + .dst_descr = "S/PDIF Output %d Playback" + }, + { + .id = 0x200, + .num = { 8, 8, 8, 4, 0 }, + .src_descr = "ADAT %d", + .src_num_offset = 1, + .dst_descr = "ADAT Output %d Playback" + }, + { + .id = 0x300, + .num = { 10, 18, 18, 18, 18 }, + .src_descr = "Mix %c", + .src_num_offset = 65, + .dst_descr = "Mixer Input %02d Capture" + }, + { + .id = 0x600, + .num = { 20, 18, 18, 14, 10 }, + .src_descr = "PCM %d", + .src_num_offset = 1, + .dst_descr = "PCM %02d Capture" + }, + }, +}; + +/* get the starting port index number for a given port type/direction */ +static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, + int direction, int port_type) +{ + int i, num = 0; + + for (i = 0; i < port_type; i++) + num += ports[i].num[direction]; + + return num; +} + +/*** USB Interactions ***/ + +/* Vendor-Specific Interface, Endpoint, MaxPacketSize, Interval */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE 5 +#define SCARLETT2_USB_INTERRUPT_ENDPOINT 4 +#define SCARLETT2_USB_INTERRUPT_MAX_DATA 64 +#define SCARLETT2_USB_INTERRUPT_INTERVAL 3 + +/* Interrupt flags for volume and mute/dim button changes */ +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 + +/* Commands for sending/receiving requests/responses */ +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 + +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA_1 0x00800001 +#define SCARLETT2_USB_SET_DATA_2 0x00800002 + +#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 +#define SCARLETT2_USB_LINE_OUT_VOLUME_OFFSET 0x34 +#define SCARLETT2_USB_SW_HW_SWITCH_OFFSET 0x66 + +#define SCARLETT2_USB_LINE_OUT_VOLUME_SET_MAGIC 1 +#define SCARLETT2_USB_SW_HW_SWITCH_SET_MAGIC 3 +#define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 + +/* Send a proprietary format request to the Scarlett interface. */ +static int scarlett2_usb( + struct usb_mixer_interface *mixer, __u32 cmd, + void *req_data, __u16 req_size, void *resp_data, __u16 resp_size) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + + /* proprietary request/response format */ + struct scarlett2_usb_packet { + __u32 cmd; + __u16 size; + __u16 seq; + __u32 error; + __u32 pad; + __u8 data[]; + }; + + __u16 req_buf_size = + sizeof(struct scarlett2_usb_packet) + req_size; + __u16 resp_buf_size = + sizeof(struct scarlett2_usb_packet) + resp_size; + + struct scarlett2_usb_packet *req, *resp; + + int seq, err; + __u32 resp_cmd; + __u32 resp_seq; + __u16 size; + + mutex_lock(&private->usb_mutex); + + /* build request message and send it */ + + req = kmalloc(req_buf_size, GFP_KERNEL); + if (!req) { + mutex_unlock(&private->usb_mutex); + return -ENOMEM; + } + + /* sequence must go up by 1 for each request */ + seq = private->scarlett2_seq++; + + req->cmd = cpu_to_le32(cmd); + req->size = cpu_to_le16(req_size); + req->seq = cpu_to_le16(seq); + req->error = 0; + req->pad = 0; + if (req_size) + memcpy(req->data, req_data, req_size); + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + req, + req_buf_size); + + kfree(req); + + if (err != req_buf_size) { + snd_printk(KERN_ERR "Scarlett Gen 2 USB request result " + "command %x was %d\n", cmd, err); + mutex_unlock(&private->usb_mutex); + return -EINVAL; + } + + /* send a second message to get the response */ + + resp = kmalloc(resp_buf_size, GFP_KERNEL); + if (!resp) { + mutex_unlock(&private->usb_mutex); + return -ENOMEM; + } + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + 0, + SCARLETT2_USB_VENDOR_SPECIFIC_INTERFACE, + resp, + resp_buf_size); + + /* validate the response */ + + if (err != resp_buf_size) { + snd_printk(KERN_ERR "Scarlett Gen 2 USB response result " + "command %x was %d\n", cmd, err); + kfree(resp); + mutex_unlock(&private->usb_mutex); + return -EINVAL; + } + + resp_cmd = le32_to_cpu(resp->cmd); + resp_seq = le16_to_cpu(resp->seq); + size = le16_to_cpu(resp->size); + + if (resp_cmd != cmd || + resp_seq != seq || + size != resp_size || + resp->error || + resp->pad) { + snd_printk(KERN_ERR "Scarlett Gen 2 USB invalid response; " + "cmd tx/rx %d/%d seq %d/%d size %d/%d " + "error %d pad %d\n", + cmd, resp_cmd, seq, resp_seq, size, resp_size, + resp->error, resp->pad); + kfree(resp); + mutex_unlock(&private->usb_mutex); + return -EINVAL; + } + + if (resp_size > 0) + memcpy(resp_data, resp->data, resp_size); + + kfree(resp); + mutex_unlock(&private->usb_mutex); + return 0; +} + +/* Send a USB message to set data (currently volume level or switch) */ +static int scarlett2_usb_set( + struct usb_mixer_interface *mixer, + int index, int size, int magic_value, int value) +{ + struct __attribute__((__packed__)) { + __u32 index; + __u32 bytes; + __s32 value; + } req; + __u32 req2; + int err; + + /* send first message with the data */ + req.index = cpu_to_le32(index); + req.bytes = cpu_to_le32(size); + req.value = cpu_to_le32(value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA_1, + &req, sizeof(__u32) * 2 + size, NULL, 0); + if (err < 0) + return err; + + /* send second message with the magic value */ + req2 = cpu_to_le32(magic_value); + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA_2, + &req2, sizeof(req2), NULL, 0); + if (err < 0) + return err; + + return 0; +} + +/* Send a USB message to get data; result placed in *buf */ +static int scarlett2_usb_get( + struct usb_mixer_interface *mixer, + __u32 cmd, int offset, void *buf, int size) +{ + struct __attribute__((__packed__)) { + __u32 offset; + __u32 size; + } req; + int err; + + /* send first message with the data */ + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + err = scarlett2_usb(mixer, cmd, &req, sizeof(req), buf, size); + if (err < 0) + return err; + + return 0; +} + +/* Send a USB message to get volume status; result placed in *buf */ +static int scarlett2_usb_get_volume_status( + struct usb_mixer_interface *mixer, + struct scarlett2_usb_volume_status *buf) +{ + return scarlett2_usb_get( + mixer, + SCARLETT2_USB_GET_DATA, + SCARLETT2_USB_VOLUME_STATUS_OFFSET, + buf, sizeof(*buf)); +} + +/* Send a USB message to set the software volume for an analogue output */ +static int scarlett2_usb_set_line_output_volume( + struct usb_mixer_interface *mixer, + int line_num, int volume) +{ + int offset = SCARLETT2_USB_LINE_OUT_VOLUME_OFFSET + line_num * 2; + + return scarlett2_usb_set( + mixer, offset, 2, + SCARLETT2_USB_LINE_OUT_VOLUME_SET_MAGIC, volume); +} + +/* Send a USB message to set software (0) or hardware (1) volume for + * an analogue output + */ +static int scarlett2_usb_set_monitor_volume_switch( + struct usb_mixer_interface *mixer, int line_num, int value) +{ + int offset = SCARLETT2_USB_SW_HW_SWITCH_OFFSET + line_num; + + return scarlett2_usb_set( + mixer, offset, 1, + SCARLETT2_USB_SW_HW_SWITCH_SET_MAGIC, value); +} + +/* Send a USB message to set the volumes for all inputs of one mix + * (values obtained from private->mix[]) + */ +static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, + int mix_num) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_device_info *info = private->info; + + struct __attribute__((__packed__)) { + __u16 mix_num; + __u16 data[SCARLETT2_INPUT_MIX_MAX]; + } req; + + int i, j, err; + int num_mixer_in = + info->ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + + req.mix_num = cpu_to_le16(mix_num); + + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + req.data[i] = cpu_to_le16( + scarlett2_mixer_values[private->mix[j]] + ); + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MIX, + &req, (num_mixer_in + 1) * sizeof(__u16), + NULL, 0); + if (err < 0) + return err; + + return 0; +} + +/* Convert a port number index (per info->ports) to a hardware ID */ +static __u32 scarlett2_mux_src_num_to_id(const struct scarlett2_ports *ports, + int num) +{ + int port_type; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (num < ports[port_type].num[SCARLETT2_PORT_IN]) + return ports[port_type].id | num; + num -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + /* Oops */ + return 0; +} + +/* Send USB messages to set mux inputs */ +static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int rate, port_dir_rate; + + static const int assignment_order[SCARLETT2_PORT_TYPE_COUNT] = { + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_NONE, + }; + + struct __attribute__((__packed__)) { + __u16 pad; + __u16 num; + __u32 data[SCARLETT2_MUX_MAX]; + } req; + + req.pad = 0; + + /* mux settings for each rate */ + for (rate = 0, port_dir_rate = SCARLETT2_PORT_OUT_44; + port_dir_rate <= SCARLETT2_PORT_OUT_176; + rate++, port_dir_rate++) { + int order_num, i, err; + + req.num = cpu_to_le16(rate); + + for (order_num = 0, i = 0; + order_num < SCARLETT2_PORT_TYPE_COUNT; + order_num++) { + int port_type = assignment_order[order_num]; + int j = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + port_type); + int port_id = ports[port_type].id; + int channel; + + for (channel = 0; + channel < ports[port_type].num[port_dir_rate]; + channel++, i++, j++) + /* lower 12 bits for the destination and + * next 12 bits for the source + */ + req.data[i] = !port_id + ? 0 + : cpu_to_le32( + port_id | + channel | + scarlett2_mux_src_num_to_id( + ports, private->mux[j] + ) << 12 + ); + + /* skip private->mux[j] entries not output */ + j += ports[port_type].num[SCARLETT2_PORT_OUT] - + ports[port_type].num[port_dir_rate]; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, + &req, (i + 1) * sizeof(__u32), + NULL, 0); + if (err < 0) + return err; + } + + return 0; +} + +/* Send USB message to get meter levels */ +static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, + __u16 *levels) +{ + struct __attribute__((__packed__)) { + __u16 pad; + __u16 num_meters; + __u32 magic; + } req; + __u32 resp[SCARLETT2_NUM_METERS]; + int i, err; + + req.pad = 0; + req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); + req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + &req, sizeof(req), resp, sizeof(resp)); + if (err < 0) + return err; + + /* copy, convert to u16 */ + for (i = 0; i < SCARLETT2_NUM_METERS; i++) + levels[i] = resp[i]; + + return 0; +} + +/*** Control Functions ***/ + +/* helper function to create a new control */ +static int add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int channels, const char *name, + struct snd_kcontrol **kctl_return) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = index; + elem->head.id = index; + elem->channels = channels; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_usb_mixer_add_control(&elem->head, kctl); + if (err < 0) + return err; + + if (kctl_return) + *kctl_return = kctl; + + return 0; +} + +/*** Analogue Line Out Volume Controls ***/ + +/* first-time initialisation of volumes */ +static int scarlett2_init_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + struct scarlett2_usb_volume_status volume_status; + int err, i; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + int volume; + + private->vol_sw_hw_switch[i] = !!volume_status.sw_hw_switch[i]; + + volume = private->vol_sw_hw_switch[i] + ? volume_status.master_vol + : volume_status.sw_vol[i]; + volume = clamp(volume + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + private->vol[i] = volume; + } + + for (i = 0; i < SCARLETT2_BUTTON_COUNT; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +/* update hardware volume controls after receiving notification that + * they have changed + */ +static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_ports *ports = private->info->ports; + struct scarlett2_usb_volume_status volume_status; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + + private->vol_updated = 0; + + err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (err < 0) + return err; + + private->master_vol = clamp( + volume_status.master_vol + SCARLETT_VOLUME_BIAS, + 0, SCARLETT_VOLUME_BIAS); + + for (i = 0; i < num_line_out; i++) { + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } + + for (i = 0; i < SCARLETT2_BUTTON_COUNT; i++) + private->buttons[i] = !!volume_status.buttons[i]; + + return 0; +} + +static int scarlett2_volume_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_VOLUME_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(elem->head.mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->master_vol; + return 0; +} + +static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + int index = elem->control; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(elem->head.mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.integer.value[0] = private->vol[index]; + return 0; +} + +static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + oval = private->vol[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) { + mutex_unlock(&private->data_mutex); + return 0; + } + + private->vol[index] = val; + err = scarlett2_usb_set_line_output_volume(elem->head.mixer, + index, + val - SCARLETT_VOLUME_BIAS); + mutex_unlock(&private->data_mutex); + return err < 0 ? err : 1; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_gain, -SCARLETT_VOLUME_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_master_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_volume_ctl_get, + .put = scarlett2_volume_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +/*** HW/SW Volume Switch Controls ***/ + +static int scarlett2_sw_hw_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "SW", "HW" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, values); +} + +static int scarlett2_sw_hw_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + + ucontrol->value.enumerated.item[0] = + private->vol_sw_hw_switch[elem->control]; + return 0; +} + +static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + oval = private->vol_sw_hw_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) { + mutex_unlock(&private->data_mutex); + return 0; + } + + private->vol_sw_hw_switch[index] = val; + + /* Change access mode to RO (hardware controlled volume) + * or RW (software controlled volume) + */ + if (val) + private->vol_ctls[index]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + else + private->vol_ctls[index]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* Reset volume to master volume */ + private->vol[index] = private->master_vol; + + /* Set SW volume to current HW volume */ + err = scarlett2_usb_set_line_output_volume( + elem->head.mixer, index, + private->master_vol - SCARLETT_VOLUME_BIAS); + if (err < 0) { + mutex_unlock(&private->data_mutex); + return err; + } + + /* Notify of RO/RW change */ + snd_ctl_notify(elem->head.mixer->chip->card, + SNDRV_CTL_EVENT_MASK_INFO, + &private->vol_ctls[index]->id); + + /* Send SW/HW switch change to the device */ + err = scarlett2_usb_set_monitor_volume_switch(elem->head.mixer, + index, val); + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_sw_hw_enum_ctl_info, + .get = scarlett2_sw_hw_enum_ctl_get, + .put = scarlett2_sw_hw_enum_ctl_put, +}; + +/*** Mute/Dim Controls ***/ + +static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + + if (private->vol_updated) { + mutex_lock(&private->data_mutex); + scarlett2_update_volumes(elem->head.mixer); + mutex_unlock(&private->data_mutex); + } + + ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; + return 0; +} + +static const struct snd_kcontrol_new scarlett2_button_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_button_ctl_get +}; + +/*** Create the volume and volume switch controls ***/ + +static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int err, i; + char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + /* Add R/O HW volume control */ + snprintf(mx, sizeof(mx), "Master HW Playback Volume"); + err = add_new_ctl(mixer, &scarlett2_master_volume_ctl, + 0, 1, mx, &private->master_vol_ctl); + if (err < 0) + return err; + + /* Add volume controls */ + for (i = 0; i < num_line_out; i++) { + + /* Fader */ + if (info->line_out_descrs[i]) + snprintf(mx, sizeof(mx), + "Line %02d (%s) Playback Volume", + i + 1, info->line_out_descrs[i]); + else + snprintf(mx, sizeof(mx), + "Line %02d Playback Volume", + i + 1); + err = add_new_ctl(mixer, &scarlett2_line_out_volume_ctl, + i, 1, mx, &private->vol_ctls[i]); + if (err < 0) + return err; + + /* Make the fader read-only if the SW/HW switch is set to HW */ + if (private->vol_sw_hw_switch[i]) + private->vol_ctls[i]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_WRITE; + + /* SW/HW Switch */ + snprintf(mx, sizeof(mx), + "Line Out %02d Volume Control Playback Enum", + i + 1); + err = add_new_ctl(mixer, &scarlett2_sw_hw_enum_ctl, + i, 1, mx, NULL); + if (err < 0) + return err; + } + + /* Add button controls */ + for (i = 0; i < SCARLETT2_BUTTON_COUNT; i++) { + err = add_new_ctl(mixer, &scarlett2_button_ctl, + i, 1, scarlett2_button_names[i], + &private->button_ctls[i]); + if (err < 0) + return err; + } + + return 0; +} + +/*** Mixer Volume Controls ***/ + +static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT_MIXER_MAX_VALUE; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + + ucontrol->value.integer.value[0] = private->mix[elem->control]; + return 0; +} + +static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_ports *ports = info->ports; + int oval, val, num_mixer_in, mix_num, err; + + mutex_lock(&private->data_mutex); + + oval = private->mix[elem->control]; + val = ucontrol->value.integer.value[0]; + num_mixer_in = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + mix_num = elem->control / num_mixer_in; + + if (oval == val) { + mutex_unlock(&private->data_mutex); + return 0; + } + + private->mix[elem->control] = val; + err = scarlett2_usb_set_mix(elem->head.mixer, mix_num); + mutex_unlock(&private->data_mutex); + return err < 0 ? err : 1; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett_mixer, + SCARLETT_MIXER_MIN_DB * 100, + SCARLETT_MIXER_MAX_DB * 100 +); + +static const struct snd_kcontrol_new scarlett2_mixer_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_mixer_ctl_get, + .put = scarlett2_mixer_ctl_put, + .private_value = SCARLETT_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett_mixer } +}; + +static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_ports *ports = private->info->ports; + int err, i, j; + int index; + char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; + int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; + + for (i = 0, index = 0; i < num_outputs; i++) { + for (j = 0; j < num_inputs; j++, index++) { + snprintf(mx, sizeof(mx), + "Mix %c Input %02d Playback Volume", + 'A' + i, j + 1); + err = add_new_ctl(mixer, &scarlett2_mixer_ctl, + index, 1, mx, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Mux Source Selection Controls ***/ + +static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + const struct scarlett2_ports *ports = private->info->ports; + unsigned int item = uinfo->value.enumerated.item; + int items = private->num_mux_srcs; + int port_type; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (item >= items) + item = uinfo->value.enumerated.item = items - 1; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + if (item < ports[port_type].num[SCARLETT2_PORT_IN]) { + sprintf(uinfo->value.enumerated.name, + ports[port_type].src_descr, + item + ports[port_type].src_num_offset); + return 0; + } + item -= ports[port_type].num[SCARLETT2_PORT_IN]; + } + + return -EINVAL; +} + +static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + + ucontrol->value.enumerated.item[0] = private->mux[elem->control]; + return 0; +} + +static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_chip_data *private = elem->head.mixer->chip->private; + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + oval = private->mux[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, private->num_mux_srcs - 1L); + + if (oval == val) { + mutex_unlock(&private->data_mutex); + return 0; + } + + private->mux[index] = val; + err = scarlett2_usb_set_mux(elem->head.mixer); + mutex_unlock(&private->data_mutex); + return err < 0 ? err : 1; +} + +static const struct snd_kcontrol_new scarlett2_mux_src_enum_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_mux_src_enum_ctl_info, + .get = scarlett2_mux_src_enum_ctl_get, + .put = scarlett2_mux_src_enum_ctl_put, +}; + +static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_ports *ports = private->info->ports; + int port_type, channel, i; + + for (i = 0, port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + for (channel = 0; + channel < ports[port_type].num[SCARLETT2_PORT_OUT]; + channel++, i++) { + int err; + char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *const descr = ports[port_type].dst_descr; + + snprintf(mx, sizeof(mx) - 5, descr, channel + 1); + strcat(mx, " Enum"); + + err = add_new_ctl(mixer, + &scarlett2_mux_src_enum_ctl, + i, 1, mx, NULL); + if (err < 0) + return err; + } + } + + return 0; +} + +/*** Meter Controls ***/ + +static int scarlett2_meter_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 4095; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + __u16 meter_levels[SCARLETT2_NUM_METERS]; + int i, err; + + err = scarlett2_usb_get_meter_levels(elem->head.mixer, meter_levels); + if (err < 0) + return err; + + for (i = 0; i < elem->channels; i++) + ucontrol->value.integer.value[i] = meter_levels[i]; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_meter_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett2_meter_ctl_info, + .get = scarlett2_meter_ctl_get +}; + +static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) +{ + return add_new_ctl(mixer, &scarlett2_meter_ctl, + 0, SCARLETT2_NUM_METERS, "Level Meter", NULL); +} + +/*** Initialisation ***/ + +static int scarlett2_count_mux_srcs(const struct scarlett2_ports *ports) +{ + int port_type, count = 0; + + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) + count += ports[port_type].num[SCARLETT2_PORT_IN]; + + return count; +} + +/* Default routing connects PCM outputs and inputs to Analogue, + * S/PDIF, then ADAT + */ +static void scarlett2_init_routing(__u8 *mux, + const struct scarlett2_ports *ports) +{ + int i, input_num, input_count, port_type; + int output_num, output_count, port_type_connect_num; + + static const int connect_order[] = { + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + -1 + }; + + /* Assign PCM inputs (routing outputs) */ + output_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_TYPE_PCM); + output_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_OUT]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + input_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_IN, port_type); + input_count = ports[port_type].num[SCARLETT2_PORT_IN]; + for (i = 0; + i < input_count && output_count; + i++, output_count--) + mux[output_num++] = input_num++; + } + + /* Assign PCM outputs (routing inputs) */ + input_num = scarlett2_get_port_start_num(ports, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_TYPE_PCM); + input_count = ports[SCARLETT2_PORT_TYPE_PCM].num[SCARLETT2_PORT_IN]; + + for (port_type = connect_order[port_type_connect_num = 0]; + port_type >= 0; + port_type = connect_order[++port_type_connect_num]) { + output_num = scarlett2_get_port_start_num( + ports, SCARLETT2_PORT_OUT, port_type); + output_count = ports[port_type].num[SCARLETT2_PORT_OUT]; + for (i = 0; + i < output_count && input_count; + i++, input_count--) + mux[output_num++] = input_num++; + } +} + +/* Initialise private data, routing, sequence number */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) +{ + struct scarlett2_chip_data *private = + kzalloc(sizeof(struct scarlett2_chip_data), GFP_KERNEL); + + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + private->info = info; + private->num_mux_srcs = scarlett2_count_mux_srcs(info->ports); + private->scarlett2_seq = 0; + mixer->chip->private = private; + + /* Setup default routing */ + scarlett2_init_routing(private->mux, info->ports); + + /* Initialise the sequence number used for the proprietary commands */ + return scarlett2_usb(mixer, SCARLETT2_USB_INIT_SEQ, + NULL, 0, NULL, 0); +} + +/* Notify on volume change */ +static void scarlett2_mixer_interrupt_vol_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + const struct scarlett2_ports *ports = private->info->ports; + int num_line_out = + ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; + int i; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + + for (i = 0; i < num_line_out; i++) { + if (!private->vol_sw_hw_switch[i]) + continue; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); + } +} + +/* Notify on button change */ +static void scarlett2_mixer_interrupt_button_change( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_chip_data *private = mixer->chip->private; + int i; + + private->vol_updated = 1; + + for (i = 0; i < SCARLETT2_BUTTON_COUNT; i++) + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->button_ctls[i]->id); +} + +/* Interrupt callback */ +static void scarlett2_mixer_interrupt(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + __u32 data; + + if (ustatus != 0) + goto requeue; + + if (len == 8) { + data = le32_to_cpu(*(__u32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); + } else { + snd_printk(KERN_ERR "scarlett mixer interrupt length %d\n", + len); + } + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } +} + +static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) +{ + void *transfer_buffer; + + if (mixer->urb) { + snd_printk(KERN_ERR "%s: mixer urb already in use!\n", + __func__); + return 0; + } + + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; + + transfer_buffer = kmalloc(SCARLETT2_USB_INTERRUPT_MAX_DATA, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb( + mixer->urb, + mixer->chip->dev, + usb_rcvintpipe(mixer->chip->dev, + SCARLETT2_USB_INTERRUPT_ENDPOINT), + transfer_buffer, + SCARLETT2_USB_INTERRUPT_MAX_DATA, + scarlett2_mixer_interrupt, + mixer, + SCARLETT2_USB_INTERRUPT_INTERVAL); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); +} + +/* Entry point */ +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +{ + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + /* Initialise private data, routing, sequence number */ + if ((err = scarlett2_init_private(mixer, info)) < 0) + return err; + + /* Read volume levels and controls from the interface */ + if ((err = scarlett2_init_volumes(mixer)) < 0) + return err; + + /* Create the analogue output volume controls */ + if ((err = scarlett2_add_line_out_ctls(mixer)) < 0) + return err; + + /* Create the input, output, and mixer mux input selections */ + if ((err = scarlett2_add_mux_enums(mixer)) < 0) + return err; + + /* Create the matrix mixer controls */ + if ((err = scarlett2_add_mixer_ctls(mixer)) < 0) + return err; + + /* Create the level meter controls */ + if ((err = scarlett2_add_meter_ctl(mixer)) < 0) + return err; + + /* Set up the interrupt polling */ + if ((err = scarlett2_mixer_status_create(mixer)) < 0) + return err; + + return 0; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h new file mode 100644 index 000000000000..52e1dad77afd --- /dev/null +++ b/sound/usb/mixer_scarlett_gen2.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __USB_MIXER_SCARLETT_GEN2_H +#define __USB_MIXER_SCARLETT_GEN2_H + +int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_GEN2_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b9faeca645fd..1357a73c3a20 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -66,6 +66,8 @@ struct snd_usb_audio { */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + + void *private; }; #define usb_audio_err(chip, fmt, args...) \

1 0

[alsa-devel] [PATCH] ASoC: fsl_esai: Support synchronous mode
by S.j. Wang 03 Apr '19

03 Apr '19

In ESAI synchronous mode, the clock is generated by Tx, So we should always set registers of Tx which relate with the bit clock and frame clock generation (TCCR, TCR, ECR), even there is only Rx is working. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..d9fcddd55c02 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -230,6 +230,21 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return -EINVAL; } + if (esai_priv->synchronous && !tx) { + switch (clk_id) { + case ESAI_HCKR_FSYS: + fsl_esai_set_dai_sysclk(dai, ESAI_HCKT_FSYS, + freq, dir); + break; + case ESAI_HCKR_EXTAL: + fsl_esai_set_dai_sysclk(dai, ESAI_HCKT_EXTAL, + freq, dir); + break; + default: + return -EINVAL; + } + } + /* Bypass divider settings if the requirement doesn't change */ if (freq == esai_priv->hck_rate[tx] && dir == esai_priv->hck_dir[tx]) return 0; @@ -537,10 +552,21 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bclk = params_rate(params) * slot_width * esai_priv->slots; - ret = fsl_esai_set_bclk(dai, tx, bclk); + ret = fsl_esai_set_bclk(dai, esai_priv->synchronous ? true : tx, bclk); if (ret) return ret; + if (esai_priv->synchronous && !tx) { + /* Use Normal mode to support monaural audio */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? + ESAI_xCR_xMOD_NETWORK : 0); + + mask = ESAI_xCR_xSWS_MASK | ESAI_xCR_PADC; + val = ESAI_xCR_xSWS(slot_width, width) | ESAI_xCR_PADC; + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, val); + } + /* Use Normal mode to support monaural audio */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? -- 1.9.1

2 4

[alsa-devel] [PATCH] ALSA: doc: my_chip has no element ioport
by Christina Quast 03 Apr '19

03 Apr '19

chip->ioport is dereferenced in two places, but the struct is defined as follows: struct mychip { struct snd_card *card; struct pci_dev *pci; unsigned long port; int irq; }; Signed-off-by: Christina Quast <cquast(a)hanoverdisplays.com> --- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 6b154dbb02cc..132f5eb9b530 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -324,7 +324,7 @@ to details explained in the following section. strcpy(card->driver, "My Chip"); strcpy(card->shortname, "My Own Chip 123"); sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); + card->shortname, chip->port, chip->irq); /* (5) */ .... /* implemented later */ @@ -437,7 +437,7 @@ Since each component can be properly freed, the single strcpy(card->driver, "My Chip"); strcpy(card->shortname, "My Own Chip 123"); sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); + card->shortname, chip->port, chip->irq); The driver field holds the minimal ID string of the chip. This is used by alsa-lib's configurator, so keep it simple but unique. Even the -- 2.20.1

2 1

[alsa-devel] alc295_fixup_chromebook function
by Kailang 03 Apr '19

03 Apr '19

Hi Takashi, You applied below patch. https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… But I think ALC225_FIXUP_HEADSET_JACK will applicate for enable for headset button. It was not only use for Chrome OS. So, ALC225_FIXUP_HEADSET_JACK and alc295_fixup_chromebook separation were be better. Intel test https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… this patch. Mic JD was also not stable. (I don't know why. It passed with Ubuntu OS.) It need to change to initial state would be better. (original was HDA_FIXUP_ACT_PRE_PROBE) +static void alc295_fixup_chromebook(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + + switch (action) { + case HDA_FIXUP_ACT_INIT: + switch (codec->core.vendor_id) { + case 0x10ec0295: + alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); + break; + case 0x10ec0236: + alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); + break; + } + break; + } +} How could I create a fix patch for this? Thanks. BR, Kailang

3 5

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: DA7219: fix build error
by Anders Roxell 03 Apr '19

03 Apr '19

When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m the following errro pops up: ../sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c: In function ‘mt8183_da7219_max98357_dev_probe’: ../sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: ‘struct snd_soc_dai_link’ has no member named ‘platform’; did you mean ‘platforms’? dai_link->platform = NULL; ^~~~~~~~ platforms Rework to use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: ebbddc75bbe8 ("ASoC: Mediatek: MT8183: Add machine driver with DA7219") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 93536659a859..1e7e8aed8707 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -410,7 +410,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- 2.20.1

2 1

[alsa-devel] [asoc:for-5.2 101/107] sound/soc/sprd/sprd-pcm-compress.c:110 sprd_platform_compr_drain_notify() warn: variable dereferenced before check 'cstream' (see line 105)
by Dan Carpenter 03 Apr '19

03 Apr '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d5e2bf9d203e4d9e08ca2e9420c6ff22ad190af commit: cce1396936ef2b347d622b4d49718818eb32029d [101/107] ASoC: sprd: Add Spreadtrum audio compress offload support smatch warnings: sound/soc/sprd/sprd-pcm-compress.c:110 sprd_platform_compr_drain_notify() warn: variable dereferenced before check 'cstream' (see line 105) # https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/commit/?i… git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout cce1396936ef2b347d622b4d49718818eb32029d vim +/cstream +110 sound/soc/sprd/sprd-pcm-compress.c cce13969 Baolin Wang 2019-03-27 98 cce13969 Baolin Wang 2019-03-27 99 static int sprd_platform_compr_trigger(struct snd_compr_stream *cstream, cce13969 Baolin Wang 2019-03-27 100 int cmd); cce13969 Baolin Wang 2019-03-27 101 cce13969 Baolin Wang 2019-03-27 102 static void sprd_platform_compr_drain_notify(void *arg) cce13969 Baolin Wang 2019-03-27 103 { cce13969 Baolin Wang 2019-03-27 104 struct snd_compr_stream *cstream = arg; cce13969 Baolin Wang 2019-03-27 @105 struct snd_compr_runtime *runtime = cstream->runtime; ^^^^^^^^^^^^^^^^ cce13969 Baolin Wang 2019-03-27 106 struct sprd_compr_stream *stream = runtime->private_data; cce13969 Baolin Wang 2019-03-27 107 cce13969 Baolin Wang 2019-03-27 108 memset(stream->info_area, 0, sizeof(struct sprd_compr_playinfo)); cce13969 Baolin Wang 2019-03-27 109 cce13969 Baolin Wang 2019-03-27 @110 if (cstream) ^^^^^^^ cce13969 Baolin Wang 2019-03-27 111 snd_compr_drain_notify(cstream); cce13969 Baolin Wang 2019-03-27 112 } cce13969 Baolin Wang 2019-03-27 113 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH resend 0/3] ASoC: Intel: Misc. fixes
by Hans de Goede 03 Apr '19

03 Apr '19

Hi Pierre-Louis, Mark, These 3 patches (originally 2 different patch-sets) seem to have fallen through the cracks, hence this resend. Pierre-Louis, can you review these 3 patches please? Regards, Hans

3 7

[alsa-devel] [PATCH] ASoC: sprd: Fix modular build
by Mark Brown 03 Apr '19

03 Apr '19

Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Suggested-by: Baolin Wang <baolin.wang(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sprd/Makefile | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile index fa1d5476b8da..e6c2606da8e2 100644 --- a/sound/soc/sprd/Makefile +++ b/sound/soc/sprd/Makefile @@ -1,4 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # Spreadtrum Audio Support -obj-$(CONFIG_SND_SOC_SPRD) += sprd-pcm-dma.o sprd-pcm-compress.o +snd-soc-sprd-platform-objs := sprd-pcm-dma.o sprd-pcm-compress.o + +obj-$(CONFIG_SND_SOC_SPRD) += snd-soc-sprd-platform.o -- 2.20.1

2 2

[alsa-devel] [RFC PATCH V2] ASoC: hdac_hdmi: add device PM dependency
by libin.yang＠intel.com 03 Apr '19

03 Apr '19

From: Libin Yang <libin.yang(a)intel.com> HDMI audio codec register operation depends on the display audio power domain. The hdmi audio codec dapm event callback must be called after display audio power is turned on. Add hdac_hdmi_device_link_add/del() in HDMI audio codec driver. The customer audio driver, such as cAVS/SST and SOF audio driver, can call the function to set/remove the audio power dependency. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 86 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hdmi.h | 7 ++++ 2 files changed, 93 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..7490219 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -133,6 +133,12 @@ struct hdac_hdmi_drv_data { int port_num; }; +/* each supplier - consumer pair has one instance of this structure */ +struct hdac_hdmi_dev_link { + struct list_head head; + struct device_link *link; +}; + struct hdac_hdmi_priv { struct hdac_device *hdev; struct snd_soc_component *component; @@ -141,10 +147,12 @@ struct hdac_hdmi_priv { struct list_head pin_list; struct list_head cvt_list; struct list_head pcm_list; + struct list_head link_list; /* supplier - consumer pairs */ int num_pin; int num_cvt; int num_ports; struct mutex pin_mutex; + struct mutex link_mutex; struct hdac_chmap chmap; struct hdac_hdmi_drv_data *drv_data; struct snd_soc_dai_driver *dai_drv; @@ -1792,6 +1800,72 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); +/** + * hdac_hdmi_device_link_add - add a link between consumer and hdev device + * @consumer: Consumer end of the link. + * @component: The hdmi codec component. + * @flags: Link flags. + * + * If there is a consumer of the hdev device, which means a consumer device + * operation depends on the hdev device power status, the consumer device + * driver should call this function. It supports several consumers depending + * on the same hdmi codec hdev device. + * + * Return: pointer of device_link, or NULL if fails to create device_link + */ +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + struct hdac_hdmi_dev_link *dev_link; + + dev_link = devm_kzalloc(&hdev->dev, sizeof(*dev_link), GFP_KERNEL); + if (!dev_link) + return NULL; + + dev_link->link = device_link_add(consumer, &hdev->dev, flags); + + mutex_lock(&hdmi->link_mutex); + if (dev_link->link) + list_add(&dev_link->head, &hdmi->link_list); + mutex_unlock(&hdmi->link_mutex); + + return dev_link->link; +} +EXPORT_SYMBOL_GPL(hdac_hdmi_device_link_add); + +/** + * hdac_hdmi_device_link_del - del the link between consumer and hdev device + * @consumer: Consumer end of the link. + * @component: The hdmi codec component. + * + * If the consumer doesn't need the dependency any longer, the consumer device + * driver should call this function to release the device_link + */ +void hdac_hdmi_device_link_del(struct device *consumer, + struct snd_soc_component *component) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + struct hdac_hdmi_dev_link *dev_link, *tmp; + + mutex_lock(&hdmi->link_mutex); + list_for_each_entry_safe(dev_link, tmp, &hdmi->link_list, head) { + if (dev_link->link->consumer == consumer && + dev_link->link->supplier == &hdev->dev) { + device_link_del(dev_link->link); + list_del(&dev_link->head); + devm_kfree(&hdev->dev, dev_link); + return; + } + } + mutex_unlock(&hdmi->link_mutex); +} +EXPORT_SYMBOL_GPL(hdac_hdmi_device_link_del); + static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { @@ -2031,7 +2105,9 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) INIT_LIST_HEAD(&hdmi_priv->pin_list); INIT_LIST_HEAD(&hdmi_priv->cvt_list); INIT_LIST_HEAD(&hdmi_priv->pcm_list); + INIT_LIST_HEAD(&hdmi_priv->link_list); mutex_init(&hdmi_priv->pin_mutex); + mutex_init(&hdmi_priv->link_mutex); /* * Turned off in the runtime_suspend during the first explicit @@ -2058,8 +2134,18 @@ static int hdac_hdmi_dev_probe(struct hdac_device *hdev) static int hdac_hdmi_dev_remove(struct hdac_device *hdev) { + struct hdac_hdmi_dev_link *dev_link, *tmp; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); + snd_hdac_display_power(hdev->bus, hdev->addr, false); + mutex_unlock(&hdmi->link_mutex); + list_for_each_entry_safe(dev_link, tmp, &hdmi->link_list, head) { + device_link_del(dev_link->link); + list_del(&dev_link->head); + } + mutex_unlock(&hdmi->link_mutex); + return 0; } diff --git a/sound/soc/codecs/hdac_hdmi.h b/sound/soc/codecs/hdac_hdmi.h index 4fa2fc9..e877167 100644 --- a/sound/soc/codecs/hdac_hdmi.h +++ b/sound/soc/codecs/hdac_hdmi.h @@ -7,4 +7,11 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int pcm, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm); + +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags); +void hdac_hdmi_device_link_del(struct device *consumer, + struct snd_soc_component *component); #endif /* __HDAC_HDMI_H__ */ -- 2.7.4

2 2

[alsa-devel] [PATCH 1/3] ASoC: wm_adsp: Check for buffer in trigger stop
by Charles Keepax 03 Apr '19

03 Apr '19

Trigger stop can be called in situations where trigger start failed and as such it can't be assumed the buffer is already attached to the compressed stream or a NULL pointer may be dereferenced. Fixes: 639e5eb3c7d6 ("ASoC: wm_adsp: Correct handling of compressed streams that restart") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0aa62b26f61cc..2da4ba2da42a8 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3952,7 +3952,8 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) } break; case SNDRV_PCM_TRIGGER_STOP: - wm_adsp_buffer_clear(compr->buf); + if (wm_adsp_compr_attached(compr)) + wm_adsp_buffer_clear(compr->buf); break; default: ret = -EINVAL; -- 2.11.0

2 5

[alsa-devel] [asoc:for-5.2 98/99] sound/soc/codecs/tlv320aic31xx.c:1267:5: sparse: symbol 'aic31xx_set_jack' was not declared. Should it be static?
by kbuild test robot 03 Apr '19

03 Apr '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 90f0202bf2e15e8933a550b7dfd439b500a76e1c commit: ebf3326cd96989a498892a1b344faee3110f7877 [98/99] ASoC: tlv320aic31xx: Add headphone/headset detection reproduce: # apt-get install sparse git checkout ebf3326cd96989a498892a1b344faee3110f7877 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tlv320aic31xx.c:1267:5: sparse: symbol 'aic31xx_set_jack' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 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: Mediatek: MT8183: TS3A227: fix build error
by Anders Roxell 03 Apr '19

03 Apr '19

When building CONFIG_SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=m the following error pops up: ../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function ‘mt8183_mt6358_ts3a227_max98357_dev_probe’: ../sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: ‘struct snd_soc_dai_link’ has no member named ‘platform’; did you mean ‘platforms’? dai_link->platform = NULL; ^~~~~~~~ platforms Rework to use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 2da56232a9e1..a50c6af5c945 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -322,7 +322,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- 2.20.1

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[alsa-devel] Applied "ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe" to the asoc tree
by Mark Brown 03 Apr '19

03 Apr '19

The patch ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe 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 52a30f3bb1b15c7d27233e9947370c1c14b5cbae Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Wed, 3 Apr 2019 12:15:52 +0800 Subject: [PATCH] ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m gcc warn this: sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c: In function mt8183_da7219_max98357_dev_probe: sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: struct snd_soc_dai_link has no member named platform; did you mean platforms? dai_link->platform = NULL; ^~~~~~~~ platforms use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 93536659a859..1e7e8aed8707 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -410,7 +410,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe
by Yue Haibing 03 Apr '19

03 Apr '19

From: YueHaibing <yuehaibing(a)huawei.com> When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m gcc warn this: sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c: In function mt8183_da7219_max98357_dev_probe: sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: struct snd_soc_dai_link has no member named platform; did you mean platforms? dai_link->platform = NULL; ^~~~~~~~ platforms use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- v2: use correct build log --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 9353665..1e7e8ae 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -410,7 +410,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- 2.7.0

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[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: Fix build error in mt8183_da7219_max98357_dev_probe
by Yue Haibing 03 Apr '19

03 Apr '19

From: YueHaibing <yuehaibing(a)huawei.com> When building CONFIG_SND_SOC_MT8183_DA7219_MAX98357A=m gcc warn this: sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function mt8183_mt6358_ts3a227_max98357_dev_probe: sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: struct snd_soc_dai_link has no member named platform; did you mean platforms? dai_link->platform = NULL; ^~~~~~~~ platforms use 'dai_link->platforms' instead of 'dai_link->platform'. Fixes: 11c0269017b2 ("ASoC: Mediatek: MT8183: Add machine driver with TS3A227") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 9353665..1e7e8ae 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -410,7 +410,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) * the "platform" will not null when probe is trying * again. It's not expected normally. */ - dai_link->platform = NULL; + dai_link->platforms = NULL; if (dai_link->platform_name) continue; -- 2.7.0

2 1

[alsa-devel] [PATCH] ASoC: soc-core: Fix probe deferral following prelink failure
by Jon Hunter 03 Apr '19

03 Apr '19

From: Jonathan Hunter <jonathanh(a)nvidia.com> Commit 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error") re-worked the clean-up of any platform pointers that may have been initialised by the function snd_soc_init_platform(). This commit missed one error path where if any of the prelinks for a soundcard failed to initialise, then these platform pointers would not be cleaned-up. This then prevents the soundcard from being initialised following a probe deferral when any of the soundcard prelinks cannot be found. Fix this by ensuring that soc_cleanup_platform() is called when initialising the soundcard prelinks fails. Fixes: 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error") Signed-off-by: Jonathan Hunter <jonathanh(a)nvidia.com> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..5a5764dba147 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2797,6 +2797,7 @@ int snd_soc_register_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, link); if (ret) { + soc_cleanup_platform(card); dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); mutex_unlock(&client_mutex); -- 1.9.1

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[alsa-devel] [asoc:for-5.2 105/107] ERROR: "sprd_platform_compr_ops" [sound/soc/sprd/sprd-pcm-dma.ko] undefined!
by kbuild test robot 03 Apr '19

03 Apr '19

Hi Wen, First bad commit (maybe != root cause): tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d5e2bf9d203e4d9e08ca2e9420c6ff22ad190af commit: c705247136a523488eac806bd357c3e5d79a7acd [105/107] ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put config: mips-allmodconfig (attached as .config) compiler: mips-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 c705247136a523488eac806bd357c3e5d79a7acd # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=mips All errors (new ones prefixed by >>): >> ERROR: "sprd_platform_compr_ops" [sound/soc/sprd/sprd-pcm-dma.ko] undefined! --- 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 v15 0/6] Media Device Allocator API
by Shuah Khan 02 Apr '19

02 Apr '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. The primary focus for testing The patch series is making sure media device is released when both drivers release the media device with a series of unbind/binds on both drivers. - both au0828 and snd-usb-aduio as built-in - both au0828 and snd-usb-aduio as modules - au0828 as module and snd-usb-aduio as built-in - au0828 as built-in and snd-usb-aduio as module Test results can be found at: https://docs.google.com/document/d/1RMF8Rwj7xHJEoOx6_K2f-REgZJ63BMeAVEf-CV-… Changes since v14: - Fixed typos and Copyright date. - Made VBI shareable with audio and video. - Tested with Hans's patch to to fix second VBi stream stepping on active VBI stream. The patch didn't fix the problem. 1. Start qv4l2 -V /dev/vbi0 Start capture - Streaming active 2. Start qv4l2 -V /dev/vbi0 Start capture - Streaming active Streaming won't start and the first stream stops. There is still a problem in au0828. I am sending the patch series with audio, video, and VBI shareable with this known issue. Changes since v13: - Minor changes to variable names and other minor changes to copyright and typos suggested by Hans Verkuil. Changes since v12: - Patch 1: Fixed prototype warns from media_dev_allocator.c. Removed dependency on find_module() by adding struct module to input args. Still need module name to pass into media_device API. - Patch 2 & 4: Update media dev allocator api calls to pass in struct module pointer. - No changes to Patches 3 & 5. - Added patch 6 with a test. It can go in separately. Changes since v11: - Patch 1: Add CONFIG_MODULES dependency in media_dev_allocator files. to fix compile errors when CONFIG_MODULES is disabled. - Patch 2, 3: No changes. - Patch 4: Fix sparse error reported by Dan Carpenter. - Patch 5: Fix warns found by Hans Verkuil. - v11 was tested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. The test plan used for testing resource sharing could serve as a regression test plan and the test results can be found at: - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (6): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies selftests: media_dev_allocator api test Documentation/media/kapi/mc-core.rst | 41 +++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 135 ++++++++ drivers/media/usb/au0828/Kconfig | 2 + drivers/media/usb/au0828/au0828-core.c | 196 ++++++++--- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 63 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 327 ++++++++++++++++++ sound/usb/media.h | 74 ++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + .../media_tests/media_dev_allocator.sh | 85 +++++ 20 files changed, 963 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h create mode 100755 tools/testing/selftests/media_tests/media_dev_allocator.sh -- 2.17.1

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Re: [alsa-devel] WARNING in port_delete
by syzbot 02 Apr '19

02 Apr '19

syzbot has bisected this bug to: commit 63d86a7e85f84b8ac3b2f394570965aedbb03787 Author: Paul E. McKenney <paulmck(a)linux.vnet.ibm.com> Date: Tue May 1 20:08:46 2018 +0000 rcu: Convert rcu_grace_period_init tracepoint to gp_seq bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=17f86d2f200000 start commit: 091a1eaa Merge branch 'akpm' git tree: upstream final crash: https://syzkaller.appspot.com/x/report.txt?x=14046d2f200000 console output: https://syzkaller.appspot.com/x/log.txt?x=10046d2f200000 kernel config: https://syzkaller.appspot.com/x/.config?x=c0af03fe452b65fb dashboard link: https://syzkaller.appspot.com/bug?extid=e4c8abb920efa77bace9 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=102e2c91400000 Reported-by: syzbot+e4c8abb920efa77bace9(a)syzkaller.appspotmail.com Fixes: 63d86a7e85f8 ("rcu: Convert rcu_grace_period_init tracepoint to gp_seq") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

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[alsa-devel] [PATCH v3 00/10] ACPI / utils: Replace leaky function
by Andy Shevchenko 02 Apr '19

02 Apr '19

The acpi_dev_get_first_match_name() is missing put_device() call and thus keeping reference counting unbalanced. In order to fix the issue introduce a new helper to convert existing users one-by-one to a better API. Since v3: - convert all existing users and drop old API (Mika) - add Hans' tag Cc: MyungJoo Ham <myungjoo.ham(a)samsung.com> Cc: Chanwoo Choi <cw00.choi(a)samsung.com> Cc: Chen-Yu Tsai <wens(a)csie.org> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Andy Shevchenko (10): ACPI / utils: Introduce acpi_dev_get_first_match_dev() helper extcon: axp288: Convert to use acpi_dev_get_first_match_dev() gpio: merrifield: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcht_da7213: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcht_es8316: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcr_rt5640: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: bytcr_rt5651: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: cht_bsw_rt5645: Convert to use acpi_dev_get_first_match_dev() ASoC: Intel: cht_bsw_rt5672: Convert to use acpi_dev_get_first_match_dev() ACPI / utils: Remove deprecated function since no user left drivers/acpi/utils.c | 16 ++++++++++------ drivers/extcon/extcon-axp288.c | 9 +++++---- drivers/gpio/gpio-merrifield.c | 18 ++++++++++++++---- include/acpi/acpi_bus.h | 4 ++-- include/linux/acpi.h | 4 ++-- sound/soc/intel/boards/bytcht_da7213.c | 9 +++++---- sound/soc/intel/boards/bytcht_es8316.c | 9 +++++---- sound/soc/intel/boards/bytcr_rt5640.c | 10 +++++----- sound/soc/intel/boards/bytcr_rt5651.c | 14 ++++++++------ sound/soc/intel/boards/cht_bsw_rt5645.c | 9 +++++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 9 +++++---- 11 files changed, 66 insertions(+), 45 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headset MIC of Acer TravelMate B114-21 with ALC233
by Jian-Hong Pan 02 Apr '19

02 Apr '19

The Acer TravelMate B114-21 laptop cannot detect and record sound from headset MIC. This patch adds the ALC233_FIXUP_ACER_HEADSET_MIC HDA verb quirk chained with ALC233_FIXUP_ASUS_MIC_NO_PRESENCE pin quirk to fix this issue. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a3fb3d4c5730..bdb2227be4eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5690,6 +5690,7 @@ enum { ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, + ALC233_FIXUP_ACER_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6713,6 +6714,15 @@ static const struct hda_fixup alc269_fixups[] = { { 0x21, 0x90170150 }, /* use as headset mic, without its own jack detect */ { } } + [ALC233_FIXUP_ACER_HEADSET_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { } + }, + .chained = true, + .chain_id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE }, }; @@ -6737,6 +6747,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- 2.20.1

3 5

[alsa-devel] alsa kernel panic
by Victor L 02 Apr '19

02 Apr '19

Hi , I have Linux panic when started play music with one of my USB audio device. first please see this logs: [ 71.684730] usbcore: registered new interface driver snd-usb-audio [ 73.835355] BUG: unable to handle kernel NULL pointer dereference at 0000000000000008 [ 73.835412] IP: [<ffffffffc0b45d3c>] snd_usb_endpoint_start+0x23c/0x2f0 [snd_usb_audio] [ 73.835414] PGD 0 [ 73.835415] [ 73.835418] Oops: 0000 [#1] PREEMPT SMP [ 73.835482] Modules linked in: snd_usb_audio snd_usbmidi_lib snd_hwdep snd_pcm_oss snd_mixer_oss snd_ pcm snd_seq_midi snd_seq_midi_event snd_rawmidi snd_seq snd_seq_device snd_timer snd soundcore coretemp iwl3945 iwlegacy i915 ehci_pci iTCO_wdt ehci_hcd pcmcia iTCO_vendor_support kvm usbcore mac80211 e1000e yenta_socket sr_mod pcmcia_rsrc cfg80211 pcmcia_core lpc_ich mfd_core drm_kms_helper cdrom rfkill ata_ge neric irqbypass ptp pps_core joydev sg pcspkr i2c_i801 drm battery i2c_algo_bit ata_piix i2c_smbus usb_c ommon shpchp rng_core ac acpi_cpufreq video button nls_utf8 cifs sha256_ssse3 cmac md4 des_generic arc4 dns_resolver fscache sunrpc loop ip_tables x_tables autofs4 ext4 crc16 jbd2 crc32c_generic fscrypto ecb glue_helper lrw gf128mul ablk_helper cryptd aes_x86_64 mbcache sd_mod ahci libahci [ 73.835486] evdev libata psmouse serio_raw scsi_mod thermal [ 73.835491] CPU: 1 PID: 1099 Comm: output:oops Not tainted 4.9.144-rt93 #1 [ 73.835492] Hardware name: LENOVO 636962M/636962M, BIOS 7IET38WW (1.19 ) 03/22/2011 [ 73.835494] task: ffff8beb97502c40 task.stack: ffffac1901068000 [ 73.835505] RIP: 0010:[<ffffffffc0b45d3c>] [<ffffffffc0b45d3c>] snd_usb_endpoint_start+0x23c/0x2f0 [ snd_usb_audio] [ 73.835507] RSP: 0000:ffffac190106bcc0 EFLAGS: 00010246 [ 73.835508] RAX: 00000000ffffffea RBX: ffff8beb96c00000 RCX: 00000000ffffffea [ 73.835509] RDX: 0000000000000000 RSI: ffffffffc0b5bf98 RDI: ffff8beb99a18000 [ 73.835511] RBP: 0000000000000002 R08: ffffffffc0b5a8d3 R09: 0000000000000000 [ 73.835512] R10: 00000000fffffe00 R11: 0000000000000200 R12: ffff8beb96c80070 [ 73.835513] R13: ffff8beb99a18000 R14: 0000000000000002 R15: 0000000000000000 [ 73.835516] FS: 00007fdfedc66700(0000) GS:ffff8bebbf300000(0000) knlGS:0000000000000000 [ 73.835518] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 73.835519] CR2: 0000000000000008 CR3: 00000000b4ab6000 CR4: 0000000000000670 [ 73.835520] Stack: [ 73.835525] ffff8bebb8888c18 ffff8beb96c00000 0000000000000000 0000000000008002 [ 73.835528] 00007fdfe40197d0 ffffffffc0b4e7cd ffff8bebb8888c18 ffff8beb974b2400 [ 73.835531] ffffffffc0b5088a ffff8beb0000ac44 ffff8bebb206d300 0000000000000000 [ 73.835531] Call Trace: [ 73.835545] [<ffffffffc0b4e7cd>] ? start_endpoints+0x3d/0x150 [snd_usb_audio] [ 73.835555] [<ffffffffc0b5088a>] ? snd_usb_pcm_prepare+0x2ea/0x520 [snd_usb_audio] [ 73.835577] [<ffffffffc0b0c176>] ? snd_pcm_sw_params_user+0x56/0x90 [snd_pcm] [ 73.835584] [<ffffffffc0b0b2d9>] ? snd_pcm_do_prepare+0x19/0x30 [snd_pcm] [ 73.835591] [<ffffffffc0b09ea6>] ? snd_pcm_action_single+0x36/0x80 [snd_pcm] [ 73.835598] [<ffffffffc0b0a311>] ? snd_pcm_action_nonatomic+0x61/0x70 [snd_pcm] [ 73.835606] [<ffffffffc0b0e219>] ? snd_pcm_common_ioctl1+0x3a9/0xc60 [snd_pcm] [ 73.835674] [<ffffffffc03a73ce>] ? ext4_file_write_iter+0x22e/0x370 [ext4] [ 73.835682] [<ffffffffc0b0ecaa>] ? snd_pcm_playback_ioctl1+0x1da/0x240 [snd_pcm] [ 73.835690] [<ffffffffb9736a38>] ? __seccomp_filter+0x78/0x2a0 [ 73.835698] [<ffffffffc0b0ed33>] ? snd_pcm_playback_ioctl+0x23/0x30 [snd_pcm] [ 73.835702] [<ffffffffb982c9c4>] ? do_vfs_ioctl+0xa4/0x620 [ 73.835706] [<ffffffffb96033b4>] ? syscall_trace_enter+0x144/0x300 [ 73.835709] [<ffffffffb982cfb4>] ? SyS_ioctl+0x74/0x80 [ 73.835711] [<ffffffffb9603c56>] ? do_syscall_64+0x76/0x100 [ 73.835717] [<ffffffffb9c38b4e>] ? entry_SYSCALL_64_after_swapgs+0x58/0xc6 [ 73.835747] Code: c0 d3 a8 b5 c0 83 f8 a6 74 15 83 f8 e0 49 c7 c0 8c a8 b5 c0 48 c7 c2 74 a8 b5 c0 4c 0f 45 c2 48 8b 13 89 c1 48 c7 c6 98 bf b5 c0 <48> 8b 7a 08 89 ea 48 81 c7 98 00 00 00 e8 e2 7b f4 f8 f0 80 63 [ 73.835757] RIP [<ffffffffc0b45d3c>] snd_usb_endpoint_start+0x23c/0x2f0 [snd_usb_audio] [ 73.835758] RSP <ffffac190106bcc0> [ 73.835759] CR2: 0000000000000008 I give you more detail, I have Debian Linux 4.9.144-rt93. I tested under Debian 4.19 and Ubuntu same panic present. Please help me to fix this problem, this device working fine under Windows. I believe this problem in Linux audio driver. I ready to provide any additional details if needs. -- Regards, Viktor L

1 2

[alsa-devel] [PATCH] ASoC: dpcm: skip missing substream while applying symmetry
by Jerome Brunet 02 Apr '19

02 Apr '19

If for any reason, the backend does not have the requested substream (like capture on a playback only backend), the BE will be skipped in dpcm_be_dai_startup(). However, dpcm_apply_symmetry() does not skip those BE and will dereference the be_substream (NULL) pointer anyway. Like in dpcm_be_dai_startup(), just skip those BE. Fixes: 906c7d690c3b ("ASoC: dpcm: Apply symmetry for DPCM") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0d5ec68a1e50..99bbd724d2a6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1899,10 +1899,15 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); - struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; int i; + /* A backend may not have the requested substream */ + if (!be_substream) + continue; + + rtd = be_substream->private_data; if (rtd->dai_link->be_hw_params_fixup) continue; -- 2.20.1

2 1

[alsa-devel] [PATCH v4 1/2] ASoC: tlv320aic31xx: Add headphone/headset detection
by Andrew F. Davis 02 Apr '19

02 Apr '19

This device can detect the insertion/removal of headphones and headsets. Enable reporting this status by enabling this interrupt and forwarding this to upper-layers if a jack has been defined. This jack definition and the resulting operation from a jack detection event must currently be defined by sound card platform code until CODEC outputs to jack mappings can be defined generically. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v3: - Rebase on broonie/for-next sound/soc/codecs/tlv320aic31xx.c | 48 +++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic31xx.h | 11 ++++++++ 2 files changed, 58 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c544a1e35f5e..a60cd8c08a14 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -25,6 +25,7 @@ #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -89,6 +90,7 @@ static bool aic31xx_volatile(struct device *dev, unsigned int reg) case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ case AIC31XX_INTRDACFLAG2: case AIC31XX_INTRADCFLAG2: + case AIC31XX_HSDETECT: return true; } return false; @@ -163,6 +165,7 @@ struct aic31xx_priv { struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + struct snd_soc_jack *jack; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1261,6 +1264,20 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } +int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + + aic31xx->jack = jack; + + /* Enable/Disable jack detection */ + regmap_write(aic31xx->regmap, AIC31XX_HSDETECT, + jack ? AIC31XX_HSD_ENABLE : 0); + + return 0; +} + static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); @@ -1301,6 +1318,7 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, + .set_jack = aic31xx_set_jack, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), @@ -1405,8 +1423,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) 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_HSPLUG) { + unsigned int val; + int status = 0; + + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); + if (ret) { + dev_err(dev, "Failed to read headset type: %d\n", ret); + goto exit; + } + + switch ((val & AIC31XX_HSD_TYPE_MASK) >> + AIC31XX_HSD_TYPE_SHIFT) { + case AIC31XX_HSD_HP: + status |= SND_JACK_HEADPHONE; + break; + case AIC31XX_HSD_HS: + status |= SND_JACK_HEADSET; + break; + default: + break; + } + + if (aic31xx->jack) + snd_soc_jack_report(aic31xx->jack, status, + AIC31XX_JACK_MASK); + } if (value & ~(AIC31XX_HPLSCDETECT | - AIC31XX_HPRSCDETECT)) + AIC31XX_HPRSCDETECT | + AIC31XX_HSPLUG)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1518,6 +1563,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_HSPLUGDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 2636f2c6bc79..9985ceed97ae 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -20,6 +20,9 @@ #define AIC31XX_MINIDSP_BIT BIT(2) #define DAC31XX_BIT BIT(3) +#define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_HEADSET) + enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, @@ -220,6 +223,14 @@ struct aic31xx_pdata { /* AIC31XX_DACMUTE */ #define AIC31XX_DACMUTE_MASK GENMASK(3, 2) +/* AIC31XX_HSDETECT */ +#define AIC31XX_HSD_ENABLE BIT(7) +#define AIC31XX_HSD_TYPE_MASK GENMASK(6, 5) +#define AIC31XX_HSD_TYPE_SHIFT 5 +#define AIC31XX_HSD_NONE 0x00 +#define AIC31XX_HSD_HP 0x01 +#define AIC31XX_HSD_HS 0x03 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- 2.21.0

2 2

[alsa-devel] Applied "ASoC: tlv320aic31xx: Add button press detection" to the asoc tree
by Mark Brown 02 Apr '19

02 Apr '19

The patch ASoC: tlv320aic31xx: Add button press detection 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 90f0202bf2e15e8933a550b7dfd439b500a76e1c Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Mon, 1 Apr 2019 10:13:29 -0500 Subject: [PATCH] ASoC: tlv320aic31xx: Add button press detection This device can optionally detect headset or microphone button presses. Add support for this by passing this event to the jack layer. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 17 +++++++++++++++-- sound/soc/codecs/tlv320aic31xx.h | 3 ++- 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index a60cd8c08a14..fa47bfcb93e9 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1423,10 +1423,21 @@ static irqreturn_t aic31xx_irq(int irq, void *data) 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_HSPLUG) { + if (value & (AIC31XX_HSPLUG | AIC31XX_BUTTONPRESS)) { unsigned int val; int status = 0; + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG2, + &val); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", + ret); + goto exit; + } + + if (val & AIC31XX_BUTTONPRESS) + status |= SND_JACK_BTN_0; + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); if (ret) { dev_err(dev, "Failed to read headset type: %d\n", ret); @@ -1451,7 +1462,8 @@ static irqreturn_t aic31xx_irq(int irq, void *data) } if (value & ~(AIC31XX_HPLSCDETECT | AIC31XX_HPRSCDETECT | - AIC31XX_HSPLUG)) + AIC31XX_HSPLUG | + AIC31XX_BUTTONPRESS)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1564,6 +1576,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, AIC31XX_HSPLUGDET | + AIC31XX_BUTTONPRESSDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 9985ceed97ae..cb024955c978 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -21,7 +21,8 @@ #define DAC31XX_BIT BIT(3) #define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ - SND_JACK_HEADSET) + SND_JACK_HEADSET | \ + SND_JACK_BTN_0) enum aic31xx_type { AIC3100 = 0, -- 2.20.1

1 0

[alsa-devel] [PATCH V2] ASoC: hdac_hdmi: add device PM dependency
by libin.yang＠intel.com 02 Apr '19

02 Apr '19

From: Libin Yang <libin.yang(a)intel.com> HDMI audio codec register operation depends on the display audio power domain. The hdmi audio codec dapm event callback must be called after display audio power is turned on. Add hdac_hdmi_device_link_add() in HDMI audio codec driver. The customer audio driver, such as cAVS/SST and SOF audio driver, can call the function to set the audio power dependency. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 ++++++++++++ sound/soc/codecs/hdac_hdmi.h | 6 ++++++ 2 files changed, 18 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..c991407 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1792,6 +1792,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + + return device_link_add(consumer, &hdev->dev, flags); +} +EXPORT_SYMBOL_GPL(hdac_hdmi_device_link_add); + static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { diff --git a/sound/soc/codecs/hdac_hdmi.h b/sound/soc/codecs/hdac_hdmi.h index 4fa2fc9..9239c6b 100644 --- a/sound/soc/codecs/hdac_hdmi.h +++ b/sound/soc/codecs/hdac_hdmi.h @@ -7,4 +7,10 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int pcm, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm); + +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags); + #endif /* __HDAC_HDMI_H__ */ -- 2.7.4

3 3

[alsa-devel] regression: bad sound card type detection in audio graph
by Arnaud Pouliquen 02 Apr '19

02 Apr '19

Hello Kuninori, We are seeing a regression while rebasing on kernel V5.1 (from v4.19). The regression is linked to your update of the audio graph to merge the support of the normal and DPCM sound cards. In our sound card (described here: https://github.com/STMicroelectronics/linux/blob/v4.19-stm32mp/arch/arm/boo…) We declare a normal sound card using the WM8994 codec: Playback: CPU-DAI (SAI2a)-> CODEC-DAI/port0 Record: CPU-DAI (SAI2B)<- CODEC-DAI/port1 ... The WM8994 codec is declared with 2 ports as it supports 2 interfaces. When we probe the sound card we are detecting a wrong dai link type because the condition if (of_get_child_count(codec_port) > 1) return true. The dai link is detected as DPCM instead of the expected normal link. I checked your last commits sent on mailing list, seems that this criteria is still valid. Please could you elaborate a little bit on this criteria for the normal/DPCM selection? What is you feeling about this limitation. Do you see a way to refine the condition to allow normal sound card with multi-port codecs. Or do we have to migrate our sound card to DPCM. Thanks in advance for your answer Regards Arnaud

2 7

[alsa-devel] [PATCH] Documentation: soundwire: Ensure that code is inside the code blocks
by Jonathan Neuschäfer 02 Apr '19

02 Apr '19

The way the document is written now, Sphinx renders empty code blocks followed by the lines that should be inside them. Unindent the code-block directives to fix this. Fixes: 502c00d9c315 ("Documentation: soundwire: fix stream.rst markup warnings") Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- Documentation/driver-api/soundwire/stream.rst | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/driver-api/soundwire/stream.rst b/Documentation/driver-api/soundwire/stream.rst index 26a6064503fd..5351bd2f34a8 100644 --- a/Documentation/driver-api/soundwire/stream.rst +++ b/Documentation/driver-api/soundwire/stream.rst @@ -201,7 +201,7 @@ Bus implements below API for allocate a stream which needs to be called once per stream. From ASoC DPCM framework, this stream state maybe linked to .startup() operation. - .. code-block:: c +.. code-block:: c int sdw_alloc_stream(char * stream_name); @@ -228,7 +228,7 @@ the respective Master(s) and Slave(s) associated with stream. These APIs can only be invoked once by respective Master(s) and Slave(s). From ASoC DPCM framework, this stream state is linked to .hw_params() operation. - .. code-block:: c +.. code-block:: c int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, @@ -274,7 +274,7 @@ Bus implements below API for PREPARE state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .prepare() operation. - .. code-block:: c +.. code-block:: c int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -304,7 +304,7 @@ Bus implements below API for ENABLE state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() start operation. - .. code-block:: c +.. code-block:: c int sdw_enable_stream(struct sdw_stream_runtime * stream); @@ -332,7 +332,7 @@ Bus implements below API for DISABLED state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() stop operation. - .. code-block:: c +.. code-block:: c int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -357,7 +357,7 @@ Bus implements below API for DEPREPARED state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() stop operation. - .. code-block:: c +.. code-block:: c int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -382,7 +382,7 @@ Bus implements below APIs for RELEASE state which needs to be called by all the Master(s) and Slave(s) associated with stream. From ASoC DPCM framework, this stream state is linked to .hw_free() operation. - .. code-block:: c +.. code-block:: c int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); @@ -395,7 +395,7 @@ stream assigned as part of ALLOCATED state. In .shutdown() the data structure maintaining stream state are freed up. - .. code-block:: c +.. code-block:: c void sdw_release_stream(struct sdw_stream_runtime * stream); -- 2.20.1

2 2

[alsa-devel] [asoc:for-5.2 92/97] sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'?
by kbuild test robot 01 Apr '19

01 Apr '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 35146467bd43704c52d1d5760af9839a10ad5b12 commit: ebbddc75bbe8d8eab64d3285b0480f5ae9232b2e [92/97] ASoC: Mediatek: MT8183: Add machine driver with DA7219 config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-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 ebbddc75bbe8d8eab64d3285b0480f5ae9232b2e # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c: In function 'mt8183_da7219_max98357_dev_probe': >> sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'? dai_link->platform = NULL; ^~~~~~~~ platforms vim +413 sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c 387 388 static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) 389 { 390 struct snd_soc_card *card = &mt8183_da7219_max98357_card; 391 struct device_node *platform_node; 392 struct snd_soc_dai_link *dai_link; 393 struct pinctrl *default_pins; 394 int ret, i; 395 396 card->dev = &pdev->dev; 397 398 platform_node = of_parse_phandle(pdev->dev.of_node, 399 "mediatek,platform", 0); 400 if (!platform_node) { 401 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); 402 return -EINVAL; 403 } 404 405 for_each_card_prelinks(card, i, dai_link) { 406 /* In the alsa soc-core, the "platform" will be 407 * allocated by devm_kzalloc if null. 408 * There is a special case that registerring 409 * sound card is failed at the first time, but 410 * the "platform" will not null when probe is trying 411 * again. It's not expected normally. 412 */ > 413 dai_link->platform = NULL; 414 415 if (dai_link->platform_name) 416 continue; 417 dai_link->platform_of_node = platform_node; 418 } 419 420 mt8183_da7219_max98357_headset_dev.codec_of_node = 421 of_parse_phandle(pdev->dev.of_node, 422 "mediatek,headset-codec", 0); 423 if (!mt8183_da7219_max98357_headset_dev.codec_of_node) { 424 dev_err(&pdev->dev, 425 "Property 'mediatek,headset-codec' missing/invalid\n"); 426 return -EINVAL; 427 } 428 429 ret = devm_snd_soc_register_card(&pdev->dev, card); 430 if (ret) { 431 dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", 432 __func__, ret); 433 return ret; 434 } 435 436 default_pins = 437 devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); 438 if (IS_ERR(default_pins)) { 439 dev_err(&pdev->dev, "%s set pins failed\n", 440 __func__); 441 return PTR_ERR(default_pins); 442 } 443 444 return ret; 445 } 446 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v14 0/6] Media Device Allocator API
by Shuah Khan 01 Apr '19

01 Apr '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. The primary focus for testing The patch series is making sure media device is released when both drivers release the media device with a series of unbind/binds on both drivers. - both au0828 and snd-usb-aduio as built-in - both au0828 and snd-usb-aduio as modules - au0828 as module and snd-usb-aduio as built-in - au0828 as built-in and snd-usb-aduio as module Test results can be found at: https://docs.google.com/document/d/1RMF8Rwj7xHJEoOx6_K2f-REgZJ63BMeAVEf-CV-… Changes since v13: - Minor changes to variable names and other minor changes to copyright and typos suggested by Hans Verkuil. Changes since v12: - Patch 1: Fixed prototype warns from media_dev_allocator.c. Removed dependency on find_module() by adding struct module to input args. Still need module name to pass into media_device API. - Patch 2 & 4: Update media dev allocator api calls to pass in struct module pointer. - No changes to Patches 3 & 5. - Added patch 6 with a test. It can go in separately. Changes since v11: - Patch 1: Add CONFIG_MODULES dependency in media_dev_allocator files. to fix compile errors when CONFIG_MODULES is disabled. - Patch 2, 3: No changes. - Patch 4: Fix sparse error reported by Dan Carpenter. - Patch 5: Fix warns found by Hans Verkuil. - v11 was tested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. The test plan used for testing resource sharing could serve as a regression test plan and the test results can be found at: - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (6): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies selftests: media_dev_allocator api test Documentation/media/kapi/mc-core.rst | 41 +++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 135 ++++++++ drivers/media/usb/au0828/Kconfig | 2 + drivers/media/usb/au0828/au0828-core.c | 195 ++++++++--- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 63 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 327 ++++++++++++++++++ sound/usb/media.h | 74 ++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + .../media_tests/media_dev_allocator.sh | 85 +++++ 20 files changed, 962 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h create mode 100755 tools/testing/selftests/media_tests/media_dev_allocator.sh -- 2.17.1

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[alsa-devel] [PATCH v4] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by Jenny TC 01 Apr '19

01 Apr '19

From: Mac Chiang <mac.chiang(a)intel.com> amplifier feedback is not modeled as being dependent on any active output. Even when there is no playback happening, parts of the graph, specifically the IV sense->speaker protection->output remains active and this prevents the DSP from entering low-power states. This patch suggest a machine driver level approach where the speaker pins are enabled/disabled dynamically depending on stream start/stop events. DPAM graph representations show the feedback loop is indeed disabled and low-power states can be reached. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Harshapriya.n <harshapriya.n(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v4: - Changed device name in in kabylake_ssp0_trigger to address a conflict with recently merged patch MAXIM_DEV0_NAME -> MAX98927_DEV0_NAME ... v3: - fixed the return logic at the end of kabylake_card_late_probe() v2: - described clearly in commit message - added snd_soc_dapm_disable_pin in kabylake_card_late_probe() --- --- sound/soc/intel/boards/kbl_da7219_max98927.c | 73 +++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 2768a57..a04a6bb 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -219,8 +219,60 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return 0; } +static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int j, ret; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + const char *name = codec_dai->component->name; + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + char pin_name[20]; + + if (strcmp(name, MAX98927_DEV0_NAME) && + strcmp(name, MAX98927_DEV1_NAME) && + strcmp(name, MAX98373_DEV0_NAME) && + strcmp(name, MAX98373_DEV1_NAME)) + continue; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to enable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to disable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + } + } + + return 0; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, + .trigger = kabylake_ssp0_trigger, }; static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -956,6 +1008,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct snd_soc_component *component = NULL; int err, i = 0; char jack_name[NAME_SIZE]; @@ -982,9 +1035,25 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) if (!component) return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); - return 0; + err = hdac_hdmi_jack_port_init(component, &card->dapm); + + if (err < 0) + return err; + + err = snd_soc_dapm_disable_pin(dapm, "Left Spk"); + if (err) { + dev_err(card->dev, "failed to disable Left Spk: %d\n", err); + return err; + } + + err = snd_soc_dapm_disable_pin(dapm, "Right Spk"); + if (err) { + dev_err(card->dev, "failed to disable Right Spk: %d\n", err); + return err; + } + + return snd_soc_dapm_sync(dapm); } /* kabylake audio machine driver for SPT + DA7219 */ -- 1.9.1

1 0

[alsa-devel] [asoc:for-5.2 90/97] sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'?
by kbuild test robot 01 Apr '19

01 Apr '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 35146467bd43704c52d1d5760af9839a10ad5b12 commit: 11c0269017b212fd47c593307d2dc3eb9713b2d0 [90/97] ASoC: Mediatek: MT8183: Add machine driver with TS3A227 config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-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 11c0269017b212fd47c593307d2dc3eb9713b2d0 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c: In function 'mt8183_mt6358_ts3a227_max98357_dev_probe': >> sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c:325:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'? dai_link->platform = NULL; ^~~~~~~~ platforms vim +325 sound/soc//mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c 298 299 static int 300 mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) 301 { 302 struct snd_soc_card *card = &mt8183_mt6358_ts3a227_max98357_card; 303 struct device_node *platform_node; 304 struct snd_soc_dai_link *dai_link; 305 struct pinctrl *default_pins; 306 int ret, i; 307 308 card->dev = &pdev->dev; 309 310 platform_node = of_parse_phandle(pdev->dev.of_node, 311 "mediatek,platform", 0); 312 if (!platform_node) { 313 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); 314 return -EINVAL; 315 } 316 317 for_each_card_prelinks(card, i, dai_link) { 318 /* In the alsa soc-core, the "platform" will be 319 * allocated by devm_kzalloc if null. 320 * There is a special case that registerring 321 * sound card is failed at the first time, but 322 * the "platform" will not null when probe is trying 323 * again. It's not expected normally. 324 */ > 325 dai_link->platform = NULL; 326 327 if (dai_link->platform_name) 328 continue; 329 dai_link->platform_of_node = platform_node; 330 } 331 332 mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node = 333 of_parse_phandle(pdev->dev.of_node, 334 "mediatek,headset-codec", 0); 335 if (!mt8183_mt6358_ts3a227_max98357_headset_dev.codec_of_node) { 336 dev_err(&pdev->dev, 337 "Property 'mediatek,headset-codec' missing/invalid\n"); 338 return -EINVAL; 339 } 340 341 ret = devm_snd_soc_register_card(&pdev->dev, card); 342 if (ret) 343 dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", 344 __func__, ret); 345 346 default_pins = 347 devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); 348 if (IS_ERR(default_pins)) { 349 dev_err(&pdev->dev, "%s set pins failed\n", 350 __func__); 351 return PTR_ERR(default_pins); 352 } 353 354 return ret; 355 } 356 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: uapi: #include <time.h> in asound.h
by Daniel Mentz 01 Apr '19

01 Apr '19

The uapi header asound.h defines types based on struct timespec. We need to #include <time.h> to get access to the definition of this struct. Previously, we encountered the following error message when building applications with a clang/bionic toolchain: kernel-headers/sound/asound.h:350:19: error: field has incomplete type 'struct timespec' struct timespec trigger_tstamp; ^ The absence of the time.h #include statement does not cause build errors with glibc, because its version of stdlib.h indirectly includes time.h. Signed-off-by: Daniel Mentz <danielmentz(a)google.com> --- include/uapi/sound/asound.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 404d4b9ffe76..df1153cea0b7 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -32,6 +32,7 @@ #ifndef __KERNEL__ #include <stdlib.h> +#include <time.h> #endif /* -- 2.21.0.392.gf8f6787159e-goog

2 1

[alsa-devel] Applied "ASoC: tlv320aic32x4: Fix Common Pins" to the asoc tree
by Mark Brown 01 Apr '19

01 Apr '19

The patch ASoC: tlv320aic32x4: Fix Common Pins 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 c63adb28f6d913310430f14c69f0a2ea55eed0cc Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Sat, 30 Mar 2019 09:02:02 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Fix Common Pins The common pins were mistakenly not added to the DAPM graph. Adding these pins will allow valid graphs to be created. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 96f1526cb258..5520044929f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -490,6 +490,8 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN2_R"), SND_SOC_DAPM_INPUT("IN3_L"), SND_SOC_DAPM_INPUT("IN3_R"), + SND_SOC_DAPM_INPUT("CM_L"), + SND_SOC_DAPM_INPUT("CM_R"), }; static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: cs42l51: add multi endpoint support
by Olivier Moysan 01 Apr '19

01 Apr '19

Support multiple endpoints on cs42L51 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: STM32MP157C-DK2 board uses CS42L51 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. >From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: cs42l51_port: port { cs42l51_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; cs42l51_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/codecs/cs42l51.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index fd2bd74024c1..80da3cd73e04 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -464,6 +464,13 @@ static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) return snd_soc_component_write(component, CS42L51_DAC_OUT_CTL, reg); } +static int cs42l51_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_dai_ops cs42l51_dai_ops = { .hw_params = cs42l51_hw_params, .set_sysclk = cs42l51_set_dai_sysclk, @@ -526,6 +533,7 @@ static const struct snd_soc_component_driver soc_component_device_cs42l51 = { .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), .dapm_routes = cs42l51_routes, .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), + .of_xlate_dai_id = cs42l51_of_xlate_dai_id, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5677-spi: Add ACPI ID
by Oder Chiou 01 Apr '19

01 Apr '19

Add the ACPI ID for the product "chromebook pixel 2015" to match the coreboot settings. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677-spi.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84501c2020c7..167a02773a0b 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -25,6 +25,7 @@ #include <linux/sysfs.h> #include <linux/clk.h> #include <linux/firmware.h> +#include <linux/acpi.h> #include "rt5677-spi.h" @@ -226,9 +227,16 @@ static int rt5677_spi_probe(struct spi_device *spi) return 0; } +static const struct acpi_device_id rt5677_spi_acpi_id[] = { + { "RT5677AA", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_spi_acpi_id); + static struct spi_driver rt5677_spi_driver = { .driver = { .name = "rt5677", + .acpi_match_table = ACPI_PTR(rt5677_spi_acpi_id), }, .probe = rt5677_spi_probe, }; -- 2.21.0

2 1

[alsa-devel] [PATCH 0/4] ASoC: Mediatek: MT8183: Add support for machine driver with DA7219 or TS3A227
by Shunli Wang 01 Apr '19

01 Apr '19

ASoC: Mediatek: MT8183: Add support for machine driver with DA7219 or TS3A227 Shunli Wang (4): ASoC: Mediatek: MT8183: Add machine driver with TS3A227 ASoC: Mediatek: Add document for mt8183-mt6358-ts3a227-max98357 ASoC: Mediatek: MT8183: Add machine driver with DA7219 ASoC: Mediatek: Add document for mt8183-da7219-max98357 .../bindings/sound/mt8183-da7219-max98357.txt | 15 + .../sound/mt8183-mt6358-ts3a227-max98357.txt | 15 + sound/soc/mediatek/Kconfig | 26 ++ sound/soc/mediatek/mt8183/Makefile | 2 + sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 472 +++++++++++++++++++++ .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 382 +++++++++++++++++ 6 files changed, 912 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8183-da7219-max98357.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt create mode 100644 sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c -- 1.9.1

2 8

[alsa-devel] [PATCH][next] ASoC: fsl: fix spelling mistake: "missign" -> "missing"
by Colin King 01 Apr '19

01 Apr '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix this. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/fsl/imx-audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 72e37ca6cfbb..7983bd339c01 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -161,7 +161,7 @@ static int imx_audmix_probe(struct platform_device *pdev) } if (!audmix_np) { - dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + dev_err(&pdev->dev, "Missing DT node for parent device.\n"); return -EINVAL; } -- 2.20.1

4 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Fix typo in comment
by Charles Keepax 01 Apr '19

01 Apr '19

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5e615fbddf9a7..0aa62b26f61cc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3263,7 +3263,7 @@ static void wm_halo_stop_core(struct wm_adsp *dsp) regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, HALO_CORE_EN, 0); - /* reset halo core with CORE_SOFT_REEST */ + /* reset halo core with CORE_SOFT_RESET */ regmap_update_bits(dsp->regmap, dsp->base + HALO_CORE_SOFT_RESET, HALO_CORE_SOFT_RESET_MASK, 1); } -- 2.11.0

2 1

[alsa-devel] Applied "ASoC: simple-card-utils: remove set but not used variable 'dai_name'" to the asoc tree
by Mark Brown 01 Apr '19

01 Apr '19

The patch ASoC: simple-card-utils: remove set but not used variable 'dai_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 35146467bd43704c52d1d5760af9839a10ad5b12 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Mon, 25 Mar 2019 03:17:40 +0000 Subject: [PATCH] ASoC: simple-card-utils: remove set but not used variable 'dai_name' Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_parse_clk': sound/soc/generic/simple-card-utils.c:164:18: warning: parameter 'dai_name' set but not used [-Wunused-but-set-parameter] It's not used since commit 0580dde59438 ("ASoC: simple-card-utils: add asoc_simple_debug_info()"), so can be removed. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Mukesh Ojha <mojha(a)codeaurora.org> 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 4ed68348f939..db1458a19985 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -173,10 +173,8 @@ int asoc_simple_parse_clk(struct device *dev, * see * soc-core.c :: snd_soc_init_multicodec() */ - if (dlc) { + if (dlc) dai_of_node = dlc->of_node; - dai_name = dlc->dai_name; - } /* * Parse dai->sysclk come from "clocks = <&xxx>" -- 2.20.1

1 0

[alsa-devel] [PATCH v3 1/2] ASoC: tlv320aic31xx: Add headphone/headset detection
by Andrew F. Davis 01 Apr '19

01 Apr '19

This device can detect the insertion/removal of headphones and headsets. Enable reporting this status by enabling this interrupt and forwarding this to upper-layers if a jack has been defined. This jack definition and the resulting operation from a jack detection event must currently be defined by sound card platform code until CODEC outputs to jack mappings can be defined generically. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v1: - Rebase Changes from v2: - Rebase sound/soc/codecs/tlv320aic31xx.c | 48 +++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic31xx.h | 11 ++++++++ 2 files changed, 58 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 6744f736501f..3e11b3d41f3d 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -25,6 +25,7 @@ #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -89,6 +90,7 @@ static bool aic31xx_volatile(struct device *dev, unsigned int reg) case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ case AIC31XX_INTRDACFLAG2: case AIC31XX_INTRADCFLAG2: + case AIC31XX_HSDETECT: return true; } return false; @@ -157,6 +159,7 @@ struct aic31xx_priv { int micbias_vg; struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct snd_soc_jack *jack; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1233,6 +1236,20 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } +int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + + aic31xx->jack = jack; + + /* Enable/Disable jack detection */ + regmap_write(aic31xx->regmap, AIC31XX_HSDETECT, + jack ? AIC31XX_HSD_ENABLE : 0); + + return 0; +} + static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); @@ -1258,6 +1275,7 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, + .set_jack = aic31xx_set_jack, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), @@ -1362,8 +1380,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) 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_HSPLUG) { + unsigned int val; + int status = 0; + + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); + if (ret) { + dev_err(dev, "Failed to read headset type: %d\n", ret); + goto exit; + } + + switch ((val & AIC31XX_HSD_TYPE_MASK) >> + AIC31XX_HSD_TYPE_SHIFT) { + case AIC31XX_HSD_HP: + status |= SND_JACK_HEADPHONE; + break; + case AIC31XX_HSD_HS: + status |= SND_JACK_HEADSET; + break; + default: + break; + } + + if (aic31xx->jack) + snd_soc_jack_report(aic31xx->jack, status, + AIC31XX_JACK_MASK); + } if (value & ~(AIC31XX_HPLSCDETECT | - AIC31XX_HPRSCDETECT)) + AIC31XX_HPRSCDETECT | + AIC31XX_HSPLUG)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1476,6 +1521,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_HSPLUGDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 2636f2c6bc79..9985ceed97ae 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -20,6 +20,9 @@ #define AIC31XX_MINIDSP_BIT BIT(2) #define DAC31XX_BIT BIT(3) +#define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_HEADSET) + enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, @@ -220,6 +223,14 @@ struct aic31xx_pdata { /* AIC31XX_DACMUTE */ #define AIC31XX_DACMUTE_MASK GENMASK(3, 2) +/* AIC31XX_HSDETECT */ +#define AIC31XX_HSD_ENABLE BIT(7) +#define AIC31XX_HSD_TYPE_MASK GENMASK(6, 5) +#define AIC31XX_HSD_TYPE_SHIFT 5 +#define AIC31XX_HSD_NONE 0x00 +#define AIC31XX_HSD_HP 0x01 +#define AIC31XX_HSD_HS 0x03 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- 2.21.0

2 2

Re: [alsa-devel] [PATCH 1/3] ASoC: cros_ec_codec: Rename ec_response_codec_gain into ec_codec_i2s_gain
by Mark Brown 01 Apr '19

01 Apr '19

On Fri, Mar 29, 2019 at 10:56:36AM -0700, Gwendal Grignou wrote: > In preparation to update cros_ec_commands.h to match ChromeOS EC code > base, rename data structure that will be merged together. > > TEST=compile. This doesn't apply against current code, please check and resend.

1 0

Re: [alsa-devel] [PATCH] max98357a:add dai without triggered by pcm
by Mark Brown 01 Apr '19

01 Apr '19

On Fri, Mar 29, 2019 at 09:51:59AM +0800, tony.zou.hq(a)gmail.com wrote: > max98357a's enable pin need setting independently > when max98357a is shared I2S with other codec. > add dai "max98357a-hifi" without pcm trigger, > and use "Spk PA Switch" to set the enable pin. This sounds like a really bad idea - if you have multiple CODECs on a single DAI link tell the core about it, don't try to bodge around it. There's support for multiple CODECs on a DAI link already. > +static const char * const ext_spk_text[] = { > + "Off", "On" > +}; Simple on/off switches should be normal switch controls, not enums.

1 0

Re: [alsa-devel] INFO: task hung in evdev_release
by syzbot 01 Apr '19

01 Apr '19

syzbot has bisected this bug to: commit e32d99af6830c9a8f37b4f2637ef0cdc60fa79fb Author: Jerome Brunet <jbrunet(a)baylibre.com> Date: Tue Jul 17 15:42:50 2018 +0000 ASoC: meson: add axg fifos DT binding documentation bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=13fba69b200000 start commit: 9dcd936c Merge tag 'for-4.19/dm-fixes-4' of git://git.kern.. git tree: upstream final crash: https://syzkaller.appspot.com/x/report.txt?x=1007a69b200000 console output: https://syzkaller.appspot.com/x/log.txt?x=17fba69b200000 kernel config: https://syzkaller.appspot.com/x/.config?x=88e9a8a39dc0be2d dashboard link: https://syzkaller.appspot.com/bug?extid=a979743610b4755d4d57 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=10fcd826400000 Reported-by: syzbot+a979743610b4755d4d57(a)syzkaller.appspotmail.com Fixes: e32d99af6830 ("ASoC: meson: add axg fifos DT binding documentation") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

2 1

[alsa-devel] [PATCH v2 0/7] ASoC: tlv320aic26: Add device tree support and bug fixes
by Cormier, Jonathan 31 Mar '19

31 Mar '19

Hi, The following patches were created to get the tlv320aic26 working on our AM335x SoM. I'm resubmitting this patch series. Previous version here: https://lore.kernel.org/patchwork/cover/601439/ This patch series relies on the following patch for proper operation. commit 93d0ad8f374c ("ASoC: tlv320aic26: Convert to params_width()") Changes since RFC: - Removed the unnecessary "#if defined(CONFIG_OF)" - Dropped the DSP_B patch Changes since V1: - Reordered Kconfig patch to first so each patch can be built - Rebased changes to Linux 5.1-rc1 Cormier, Jonathan (7): ASoC: tlv320aic26: Allow tlv320aic26 to be selected in menuconfig dt-bindings: ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Fix module autoload ASoC: tlv320aic26: Fix regmap by setting reg_defaults and reg_stride ASoC: tlv320aic26: hw_params was unintentionally clearing AIC26 master mode ASoC: tlv320aic26: Change Capture Mute to Capture Switch to match alsa documentation. .../devicetree/bindings/sound/tlv320aic26.txt | 65 +++++++++++++++++++ sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/tlv320aic26.c | 55 ++++++++++++++-- 3 files changed, 116 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic26.txt -- 2.21.0

2 8

[alsa-devel] [PATCH] ASoC: hdmi: add device PM dependency
by libin.yang＠intel.com 31 Mar '19

31 Mar '19

From: Libin Yang <libin.yang(a)intel.com> HDMI audio codec register operation depends on the display audio power domain. The hdmi audio codec dapm event callback must be called after display audio power is turned on. Add hdac_hdmi_device_link_add() in HDMI audio codec driver. The customer audio driver can call the function to set the audio power dependency. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 ++++++++++++ sound/soc/codecs/hdac_hdmi.h | 6 ++++++ 2 files changed, 18 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..c991407 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1792,6 +1792,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + + return device_link_add(consumer, &hdev->dev, flags); +} +EXPORT_SYMBOL_GPL(hdac_hdmi_device_link_add); + static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { diff --git a/sound/soc/codecs/hdac_hdmi.h b/sound/soc/codecs/hdac_hdmi.h index 4fa2fc9..9239c6b 100644 --- a/sound/soc/codecs/hdac_hdmi.h +++ b/sound/soc/codecs/hdac_hdmi.h @@ -7,4 +7,10 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int pcm, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm); + +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags); + #endif /* __HDAC_HDMI_H__ */ -- 2.7.4

3 2

Re: [alsa-devel] [PATCH v3 2/3] mfd: cros: Update EC protocol to match current EC code
by Jonathan Cameron 30 Mar '19

30 Mar '19

On Fri, 29 Mar 2019 10:56:36 -0700 Gwendal Grignou <gwendal(a)chromium.org> wrote: > Chromebook Embedded Controller protocol is defined in the kernel at > cros_ec_commands.h. > The source of trust for the EC protocol is at > https://chromium.googlesource.com/chromiumos/platform/ec/+/master/include/e… > > Only needed changes have been picked up from this file to the kernel > include file leading to gaps between the upstream version and what the > latest ECs can do. > > Fill the gaps to ease future integrations. Changes from the original > files is header/footer for license and include files for alignment. > > Check this include file works on ChomeOS kernel 4.14 and 4.19 on eve. > > Signed-off-by: Gwendal Grignou <gwendal(a)chromium.org> > Acked-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> I think I'd personally prefer to see this as multiple patches. 1. Functional changes to existing elements, including cases where documentation is updated to reflect the true state. 2. Trivial formatting changes or non functional comment changes. 3. SPDX change 4. New stuff. Probably as multiple patches to make them easier to review. 5. Compile time trickery (to separate the discussion around that). > --- > Changes in v3: > - Going back to v1, patch not processed properly by patchwork > > Changes in v2: > - Replace 1 << with BIT() > - Check checkpatch warning/errors are outside kernel scope. > - Update .h with latest changes. > include/linux/mfd/cros_ec_commands.h | 3627 +++++++++++++++++++++----- > 1 file changed, 2935 insertions(+), 692 deletions(-) > > diff --git a/include/linux/mfd/cros_ec_commands.h > b/include/linux/mfd/cros_ec_commands.h > index c9b58b1133e9..ea25e1644d23 100644 > --- a/include/linux/mfd/cros_ec_commands.h > +++ b/include/linux/mfd/cros_ec_commands.h > @@ -1,28 +1,41 @@ > +/* SPDX-License-Identifier: GPL-2.0 */ > /* > * Host communication command constants for ChromeOS EC > * > * Copyright (C) 2012 Google, Inc > * > - * This software is licensed under the terms of the GNU General Public > - * License version 2, as published by the Free Software Foundation, and > - * may be copied, distributed, and modified under those terms. > - * > - * This program is distributed in the hope that it will be useful, > - * but WITHOUT ANY WARRANTY; without even the implied warranty of > - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the > - * GNU General Public License for more details. > - * > - * The ChromeOS EC multi function device is used to mux all the requests > - * to the EC device for its multiple features: keyboard controller, > - * battery charging and regulator control, firmware update. > - * > * NOTE: This file is copied verbatim from the ChromeOS EC Open Source > * project in an attempt to make future updates easy to make. > */ > > +/* Host communication command constants for Chrome EC */ > + > #ifndef __CROS_EC_COMMANDS_H > #define __CROS_EC_COMMANDS_H > > +#if !defined(__ACPI__) && !defined(__KERNEL__) > +#include <stdint.h> > +#endif > + > +#ifdef __cplusplus > +extern "C"{ > +#endif I'm going to stand back on these. Personal feeling is we don't want them in kernel source. It might make future integration easier but it also adds stuff that is of no interest to us at all. However, it seems there is already some of this in the tree for this driver so I guess people don't really mind. > + > +/* > + * Include common.h for CONFIG_HOSTCMD_ALIGNED, if it's defined. This > + * generates more efficient code for accessing request/response structures on > + * ARM Cortex-M if the structures are guaranteed 32-bit aligned. > + */ > +#ifdef CHROMIUM_EC > +#include "common.h" > +#endif > + > +#ifdef __KERNEL__ > +#define BUILD_ASSERT(_cond) > +#else > +#include "compile_time_macros.h" > +#endif > + > /* > * Current version of this protocol > * > @@ -85,13 +98,15 @@ > /* Unused 0x28 - 0x2f */ > #define EC_MEMMAP_SWITCHES 0x30 /* 8 bits */ > /* Unused 0x31 - 0x33 */ > -#define EC_MEMMAP_HOST_EVENTS 0x34 /* 32 bits */ > -/* Reserve 0x38 - 0x3f for additional host event-related stuff */ > -/* Battery values are all 32 bits */ > +#define EC_MEMMAP_HOST_EVENTS 0x34 /* 64 bits */ > +/* Battery values are all 32 bits, unless otherwise noted. */ > #define EC_MEMMAP_BATT_VOLT 0x40 /* Battery Present Voltage */ > #define EC_MEMMAP_BATT_RATE 0x44 /* Battery Present Rate */ > #define EC_MEMMAP_BATT_CAP 0x48 /* Battery Remaining Capacity */ > -#define EC_MEMMAP_BATT_FLAG 0x4c /* Battery State, defined below */ > +#define EC_MEMMAP_BATT_FLAG 0x4c /* Battery State, see below (8-bit) */ > +#define EC_MEMMAP_BATT_COUNT 0x4d /* Battery Count (8-bit) */ > +#define EC_MEMMAP_BATT_INDEX 0x4e /* Current Battery Data Index > (8-bit) */ > +/* Unused 0x4f */ > #define EC_MEMMAP_BATT_DCAP 0x50 /* Battery Design Capacity */ > #define EC_MEMMAP_BATT_DVLT 0x54 /* Battery Design Voltage */ > #define EC_MEMMAP_BATT_LFCC 0x58 /* Battery Last Full Charge Capacity */ > @@ -105,10 +120,19 @@ > /* Unused 0x84 - 0x8f */ > #define EC_MEMMAP_ACC_STATUS 0x90 /* Accelerometer status (8 bits )*/ > /* Unused 0x91 */ > -#define EC_MEMMAP_ACC_DATA 0x92 /* Accelerometer data 0x92 - 0x9f */ > +#define EC_MEMMAP_ACC_DATA 0x92 /* Accelerometers data 0x92 - 0x9f */ > +/* 0x92: Lid Angle if available, LID_ANGLE_UNRELIABLE otherwise */ > +/* 0x94 - 0x99: 1st Accelerometer */ > +/* 0x9a - 0x9f: 2nd Accelerometer */ > #define EC_MEMMAP_GYRO_DATA 0xa0 /* Gyroscope data 0xa0 - 0xa5 */ > -/* Unused 0xa6 - 0xfe (remember, 0xff is NOT part of the memmap region) */ > +/* Unused 0xa6 - 0xdf */ > > +/* > + * ACPI is unable to access memory mapped data at or above this offset due to > + * limitations of the ACPI protocol. Do not place data in the range 0xe0 - 0xfe > + * which might be needed by ACPI. > + */ > +#define EC_MEMMAP_NO_ACPI 0xe0 > > /* Define the format of the accelerometer mapped memory status byte. */ > #define EC_MEMMAP_ACC_STATUS_SAMPLE_ID_MASK 0x0f > @@ -157,6 +181,8 @@ > #define EC_BATT_FLAG_DISCHARGING 0x04 > #define EC_BATT_FLAG_CHARGING 0x08 > #define EC_BATT_FLAG_LEVEL_CRITICAL 0x10 > +/* Set if some of the static/dynamic data is invalid (or outdated). */ > +#define EC_BATT_FLAG_INVALID_DATA 0x20 > > /* Switch flags at EC_MEMMAP_SWITCHES */ > #define EC_SWITCH_LID_OPEN 0x01 > @@ -182,6 +208,194 @@ > #define EC_WIRELESS_SWITCH_WWAN 0x04 /* WWAN power */ > #define EC_WIRELESS_SWITCH_WLAN_POWER 0x08 /* WLAN power */ > > +/*****************************************************************************/ > +/* > + * ACPI commands > + * > + * These are valid ONLY on the ACPI command/data port. > + */ Is this relevant to the kernel? > + > +/* > + * ACPI Read Embedded Controller > + * > + * This reads from ACPI memory space on the EC (EC_ACPI_MEM_*). > + * > + * Use the following sequence: > + * > + * - Write EC_CMD_ACPI_READ to EC_LPC_ADDR_ACPI_CMD > + * - Wait for EC_LPC_CMDR_PENDING bit to clear > + * - Write address to EC_LPC_ADDR_ACPI_DATA > + * - Wait for EC_LPC_CMDR_DATA bit to set > + * - Read value from EC_LPC_ADDR_ACPI_DATA > + */ > +#define EC_CMD_ACPI_READ 0x0080 > + > +/* > + * ACPI Write Embedded Controller > + * > + * This reads from ACPI memory space on the EC (EC_ACPI_MEM_*). > + * > + * Use the following sequence: > + * > + * - Write EC_CMD_ACPI_WRITE to EC_LPC_ADDR_ACPI_CMD > + * - Wait for EC_LPC_CMDR_PENDING bit to clear > + * - Write address to EC_LPC_ADDR_ACPI_DATA > + * - Wait for EC_LPC_CMDR_PENDING bit to clear > + * - Write value to EC_LPC_ADDR_ACPI_DATA > + */ > +#define EC_CMD_ACPI_WRITE 0x0081 > + > +/* > + * ACPI Burst Enable Embedded Controller > + * > + * This enables burst mode on the EC to allow the host to issue several > + * commands back-to-back. While in this mode, writes to mapped multi-byte > + * data are locked out to ensure data consistency. > + */ > +#define EC_CMD_ACPI_BURST_ENABLE 0x0082 > + > +/* > + * ACPI Burst Disable Embedded Controller > + * > + * This disables burst mode on the EC and stops preventing EC writes to mapped > + * multi-byte data. > + */ > +#define EC_CMD_ACPI_BURST_DISABLE 0x0083 > + > +/* > + * ACPI Query Embedded Controller > + * > + * This clears the lowest-order bit in the currently pending host events, and > + * sets the result code to the 1-based index of the bit (event 0x00000001 = 1, > + * event 0x80000000 = 32), or 0 if no event was pending. > + */ > +#define EC_CMD_ACPI_QUERY_EVENT 0x0084 > + > +/* Valid addresses in ACPI memory space, for read/write commands */ > + > +/* Memory space version; set to EC_ACPI_MEM_VERSION_CURRENT */ > +#define EC_ACPI_MEM_VERSION 0x00 > +/* > + * Test location; writing value here updates test compliment byte to (0xff - > + * value). > + */ > +#define EC_ACPI_MEM_TEST 0x01 > +/* Test compliment; writes here are ignored. */ > +#define EC_ACPI_MEM_TEST_COMPLIMENT 0x02 > + > +/* Keyboard backlight brightness percent (0 - 100) */ > +#define EC_ACPI_MEM_KEYBOARD_BACKLIGHT 0x03 > +/* DPTF Target Fan Duty (0-100, 0xff for auto/none) */ > +#define EC_ACPI_MEM_FAN_DUTY 0x04 > + > +/* > + * DPTF temp thresholds. Any of the EC's temp sensors can have up to two > + * independent thresholds attached to them. The current value of the ID > + * register determines which sensor is affected by the THRESHOLD and COMMIT > + * registers. The THRESHOLD register uses the same EC_TEMP_SENSOR_OFFSET scheme > + * as the memory-mapped sensors. The COMMIT register applies those settings. > + * > + * The spec does not mandate any way to read back the threshold settings > + * themselves, but when a threshold is crossed the AP needs a way to determine > + * which sensor(s) are responsible. Each reading of the ID register clears and > + * returns one sensor ID that has crossed one of its threshold (in either > + * direction) since the last read. A value of 0xFF means "no new thresholds > + * have tripped". Setting or enabling the thresholds for a sensor will clear > + * the unread event count for that sensor. > + */ > +#define EC_ACPI_MEM_TEMP_ID 0x05 > +#define EC_ACPI_MEM_TEMP_THRESHOLD 0x06 > +#define EC_ACPI_MEM_TEMP_COMMIT 0x07 > +/* > + * Here are the bits for the COMMIT register: > + * bit 0 selects the threshold index for the chosen sensor (0/1) > + * bit 1 enables/disables the selected threshold (0 = off, 1 = on) > + * Each write to the commit register affects one threshold. > + */ > +#define EC_ACPI_MEM_TEMP_COMMIT_SELECT_MASK (1 << 0) > +#define EC_ACPI_MEM_TEMP_COMMIT_ENABLE_MASK (1 << 1) Nice to bring these in directly as BIT() > +/* > + * Example: > + * > + * Set the thresholds for sensor 2 to 50 C and 60 C: > + * write 2 to [0x05] -- select temp sensor 2 > + * write 0x7b to [0x06] -- C_TO_K(50) - EC_TEMP_SENSOR_OFFSET > + * write 0x2 to [0x07] -- enable threshold 0 with this value > + * write 0x85 to [0x06] -- C_TO_K(60) - EC_TEMP_SENSOR_OFFSET > + * write 0x3 to [0x07] -- enable threshold 1 with this value > + * > + * Disable the 60 C threshold, leaving the 50 C threshold unchanged: > + * write 2 to [0x05] -- select temp sensor 2 > + * write 0x1 to [0x07] -- disable threshold 1 > + */ > + > +/* DPTF battery charging current limit */ > +#define EC_ACPI_MEM_CHARGING_LIMIT 0x08 > + > +/* Charging limit is specified in 64 mA steps */ > +#define EC_ACPI_MEM_CHARGING_LIMIT_STEP_MA 64 > +/* Value to disable DPTF battery charging limit */ > +#define EC_ACPI_MEM_CHARGING_LIMIT_DISABLED 0xff > + > +/* > + * Report device orientation > + * Bits Definition > + * 3:1 Device DPTF Profile Number (DDPN) > + * 0 = Reserved for backward compatibility (indicates no valid > + * profile number. Host should fall back to using TBMD). > + * 1..7 = DPTF Profile number to indicate to host which > table needs > + * to be loaded. > + * 0 Tablet Mode Device Indicator (TBMD) > + */ > +#define EC_ACPI_MEM_DEVICE_ORIENTATION 0x09 > +#define EC_ACPI_MEM_TBMD_SHIFT 0 > +#define EC_ACPI_MEM_TBMD_MASK 0x1 > +#define EC_ACPI_MEM_DDPN_SHIFT 1 > +#define EC_ACPI_MEM_DDPN_MASK 0x7 > + > +/* > + * Report device features. Uses the same format as the host command, except: > + * > + * bit 0 (EC_FEATURE_LIMITED) changes meaning from "EC code has a limited set > + * of features", which is of limited interest when the system is already > + * interpreting ACPI bytecode, to "EC_FEATURES[0-7] is not supported". Since > + * these are supported, it defaults to 0. > + * This allows detecting the presence of this field since older versions of > + * the EC codebase would simply return 0xff to that unknown address. Check > + * FEATURES0 != 0xff (or FEATURES0[0] == 0) to make sure that the other bits > + * are valid. > + */ > +#define EC_ACPI_MEM_DEVICE_FEATURES0 0x0a > +#define EC_ACPI_MEM_DEVICE_FEATURES1 0x0b > +#define EC_ACPI_MEM_DEVICE_FEATURES2 0x0c > +#define EC_ACPI_MEM_DEVICE_FEATURES3 0x0d > +#define EC_ACPI_MEM_DEVICE_FEATURES4 0x0e > +#define EC_ACPI_MEM_DEVICE_FEATURES5 0x0f > +#define EC_ACPI_MEM_DEVICE_FEATURES6 0x10 > +#define EC_ACPI_MEM_DEVICE_FEATURES7 0x11 > + > +#define EC_ACPI_MEM_BATTERY_INDEX 0x12 > + > +/* > + * USB Port Power. Each bit indicates whether the corresponding USB > ports' power > + * is enabled (1) or disabled (0). > + * bit 0 USB port ID 0 > + * ... > + * bit 7 USB port ID 7 > + */ > +#define EC_ACPI_MEM_USB_PORT_POWER 0x13 > + > +/* > + * ACPI addresses 0x20 - 0xff map to EC_MEMMAP offset 0x00 - 0xdf. This data > + * is read-only from the AP. Added in EC_ACPI_MEM_VERSION 2. > + */ > +#define EC_ACPI_MEM_MAPPED_BEGIN 0x20 > +#define EC_ACPI_MEM_MAPPED_SIZE 0xe0 > + > +/* Current version of ACPI memory address space */ > +#define EC_ACPI_MEM_VERSION_CURRENT 2 > + > + > /* > * This header file is used in coreboot both in C and ACPI code. The ACPI code > * is pre-processed to handle constants but the ASL compiler is unable to > @@ -197,6 +411,92 @@ > #define __packed __attribute__((packed)) > #endif > > +#ifndef __aligned > +#define __aligned(x) __attribute__((aligned(x))) > +#endif > + > +/* > + * Attributes for EC request and response packets. Just defining __packed > + * results in inefficient assembly code on ARM, if the structure is actually > + * 32-bit aligned, as it should be for all buffers. > + * > + * Be very careful when adding these to existing structures. They will round > + * up the structure size to the specified boundary. > + * > + * Also be very careful to make that if a structure is included in some other > + * parent structure that the alignment will still be true given the packing of > + * the parent structure. This is particularly important if the sub-structure > + * will be passed as a pointer to another function, since that function will > + * not know about the misaligment caused by the parent structure's packing. > + * > + * Also be very careful using __packed - particularly when nesting non-packed > + * structures inside packed ones. In fact, DO NOT use __packed directly; > + * always use one of these attributes. > + * > + * Once everything is annotated properly, the following search strings should > + * not return ANY matches in this file other than right here: > + * > + * "__packed" - generates inefficient code; all sub-structs must also be packed > + * > + * "struct [^_]" - all structs should be annotated, except for structs that are > + * members of other structs/unions (and their original declarations should be > + * annotated). > + */ > +#ifdef CONFIG_HOSTCMD_ALIGNED > + > +/* > + * Packed structures where offset and size are always aligned to 1, 2, or 4 > + * byte boundary. > + */ > +#define __ec_align1 __packed > +#define __ec_align2 __packed __aligned(2) > +#define __ec_align4 __packed __aligned(4) > + > +/* > + * Packed structure which must be under-aligned, because its size is not a > + * 4-byte multiple. This is sub-optimal because it forces byte-wise access > + * of all multi-byte fields in it, even though they are themselves aligned. > + * > + * In theory, we could duplicate the structure with __aligned(4) for accessing > + * its members, but use the __packed version for sizeof(). > + */ > +#define __ec_align_size1 __packed > + > +/* > + * Packed structure which must be under-aligned, because its offset inside a > + * parent structure is not a 4-byte multiple. > + */ > +#define __ec_align_offset1 __packed > +#define __ec_align_offset2 __packed __aligned(2) > + > +/* > + * Structures which are complicated enough that I'm skipping them on the first > + * pass. They are effectively unchanged from their previous definitions. > + * > + * TODO(rspangler): Figure out what to do with these. It's likely necessary > + * to work out the size and offset of each member and add explicit padding to > + * maintain those. > + */ > +#define __ec_todo_packed __packed > +#define __ec_todo_unpacked > + > +#else /* !CONFIG_HOSTCMD_ALIGNED */ > + > +/* > + * Packed structures make no assumption about alignment, so they do inefficient > + * byte-wise reads. > + */ > +#define __ec_align1 __packed > +#define __ec_align2 __packed > +#define __ec_align4 __packed > +#define __ec_align_size1 __packed > +#define __ec_align_offset1 __packed > +#define __ec_align_offset2 __packed > +#define __ec_todo_packed __packed > +#define __ec_todo_unpacked > + > +#endif /* !CONFIG_HOSTCMD_ALIGNED */ > + > /* LPC command status byte masks */ > /* EC has written a byte in the data register and host hasn't read it yet */ > #define EC_LPC_STATUS_TO_HOST 0x01 > @@ -206,7 +506,7 @@ > #define EC_LPC_STATUS_PROCESSING 0x04 > /* Last write to EC was a command, not data */ > #define EC_LPC_STATUS_LAST_CMD 0x08 > -/* EC is in burst mode. Unsupported by Chrome EC, so this bit is never set */ > +/* EC is in burst mode */ > #define EC_LPC_STATUS_BURST_MODE 0x10 > /* SCI event is pending (requesting SCI query) */ > #define EC_LPC_STATUS_SCI_PENDING 0x20 > @@ -222,7 +522,9 @@ > #define EC_LPC_STATUS_BUSY_MASK \ > (EC_LPC_STATUS_FROM_HOST | EC_LPC_STATUS_PROCESSING) > > -/* Host command response codes */ > +/* Host command response codes (16-bit). Note that response codes should be > + * stored in a uint16_t rather than directly in a value of this type. > + */ > enum ec_status { > EC_RES_SUCCESS = 0, > EC_RES_INVALID_COMMAND = 1, > @@ -238,7 +540,13 @@ enum ec_status { > EC_RES_OVERFLOW = 11, /* Table / data overflow */ > EC_RES_INVALID_HEADER = 12, /* Header contains invalid data */ > EC_RES_REQUEST_TRUNCATED = 13, /* Didn't get the entire request */ > - EC_RES_RESPONSE_TOO_BIG = 14 /* Response was too big to handle */ > + EC_RES_RESPONSE_TOO_BIG = 14, /* Response was too big to handle */ > + EC_RES_BUS_ERROR = 15, /* Communications bus error */ > + EC_RES_BUSY = 16, /* Up but too busy. Should retry */ > + EC_RES_INVALID_HEADER_VERSION = 17, /* Header version invalid */ > + EC_RES_INVALID_HEADER_CRC = 18, /* Header CRC invalid */ > + EC_RES_INVALID_DATA_CRC = 19, /* Data CRC invalid */ > + EC_RES_DUP_UNAVAILABLE = 20, /* Can't resend response */ > }; > > /* > @@ -258,7 +566,8 @@ enum host_event_code { > EC_HOST_EVENT_BATTERY_CRITICAL = 7, > EC_HOST_EVENT_BATTERY = 8, > EC_HOST_EVENT_THERMAL_THRESHOLD = 9, > - EC_HOST_EVENT_THERMAL_OVERLOAD = 10, > + /* Event generated by a device attached to the EC */ > + EC_HOST_EVENT_DEVICE = 10, > EC_HOST_EVENT_THERMAL = 11, > EC_HOST_EVENT_USB_CHARGER = 12, > EC_HOST_EVENT_KEY_PRESSED = 13, > @@ -285,15 +594,34 @@ enum host_event_code { > EC_HOST_EVENT_HANG_DETECT = 20, > /* Hang detect logic detected a hang and warm rebooted the AP */ > EC_HOST_EVENT_HANG_REBOOT = 21, > + > /* PD MCU triggering host event */ > EC_HOST_EVENT_PD_MCU = 22, > > - /* EC desires to change state of host-controlled USB mux */ > - EC_HOST_EVENT_USB_MUX = 28, > + /* Battery Status flags have changed */ > + EC_HOST_EVENT_BATTERY_STATUS = 23, > + > + /* EC encountered a panic, triggering a reset */ > + EC_HOST_EVENT_PANIC = 24, > + > + /* Keyboard fastboot combo has been pressed */ > + EC_HOST_EVENT_KEYBOARD_FASTBOOT = 25, > > /* EC RTC event occurred */ > EC_HOST_EVENT_RTC = 26, > > + /* Emulate MKBP event */ > + EC_HOST_EVENT_MKBP = 27, > + > + /* EC desires to change state of host-controlled USB mux */ > + EC_HOST_EVENT_USB_MUX = 28, > + > + /* TABLET/LAPTOP mode or detachable base attach/detach event */ > + EC_HOST_EVENT_MODE_CHANGE = 29, > + > + /* Keyboard recovery combo with hardware reinitialization */ > + EC_HOST_EVENT_KEYBOARD_RECOVERY_HW_REINIT = 30, > + > /* > * The high bit of the event mask is not used as a host event code. If > * it reads back as set, then the entire event mask should be > @@ -304,7 +632,7 @@ enum host_event_code { > EC_HOST_EVENT_INVALID = 32 > }; > /* Host event mask */ > -#define EC_HOST_EVENT_MASK(event_code) (1UL << ((event_code) - 1)) > +#define EC_HOST_EVENT_MASK(event_code) (1ULL << ((event_code) - 1)) > > /** > * struct ec_lpc_host_args - Arguments at EC_LPC_ADDR_HOST_ARGS > @@ -319,7 +647,7 @@ struct ec_lpc_host_args { > uint8_t command_version; > uint8_t data_size; > uint8_t checksum; > -} __packed; > +} __ec_align4; > > /* Flags for ec_lpc_host_args.flags */ > /* > @@ -329,7 +657,7 @@ struct ec_lpc_host_args { > * If EC gets a command and this flag is not set, this is an old-style command. > * Command version is 0 and params from host are at EC_LPC_ADDR_OLD_PARAM with > * unknown length. EC must respond with an old-style response (that is, > - * withouth setting EC_HOST_ARGS_FLAG_TO_HOST). > + * without setting EC_HOST_ARGS_FLAG_TO_HOST). > */ > #define EC_HOST_ARGS_FLAG_FROM_HOST 0x01 > /* > @@ -490,7 +818,7 @@ struct ec_host_request { > uint8_t command_version; > uint8_t reserved; > uint16_t data_len; > -} __packed; > +} __ec_align4; > > #define EC_HOST_RESPONSE_VERSION 3 > > @@ -509,18 +837,151 @@ struct ec_host_response { > uint16_t result; > uint16_t data_len; > uint16_t reserved; > -} __packed; > +} __ec_align4; > + > +/*****************************************************************************/ > + > +/* > + * Host command protocol V4. > + * > + * Packets always start with a request or response header. They are followed > + * by data_len bytes of data. If the data_crc_present flag is set, the data > + * bytes are followed by a CRC-8 of that data, using using x^8 + x^2 + x + 1 > + * polynomial. > + * > + * Host algorithm when sending a request q: > + * > + * 101) tries_left=(some value, e.g. 3); > + * 102) q.seq_num++ > + * 103) q.seq_dup=0 > + * 104) Calculate q.header_crc. > + * 105) Send request q to EC. > + * 106) Wait for response r. Go to 201 if received or 301 if timeout. > + * > + * 201) If r.struct_version != 4, go to 301. > + * 202) If r.header_crc mismatches calculated CRC for r header, go to 301. > + * 203) If r.data_crc_present and r.data_crc mismatches, go to 301. > + * 204) If r.seq_num != q.seq_num, go to 301. > + * 205) If r.seq_dup == q.seq_dup, return success. > + * 207) If r.seq_dup == 1, go to 301. > + * 208) Return error. > + * > + * 301) If --tries_left <= 0, return error. > + * 302) If q.seq_dup == 1, go to 105. > + * 303) q.seq_dup = 1 > + * 304) Go to 104. > + * > + * EC algorithm when receiving a request q. > + * EC has response buffer r, error buffer e. > + * > + * 101) If q.struct_version != 4, set e.result = EC_RES_INVALID_HEADER_VERSION > + * and go to 301 > + * 102) If q.header_crc mismatches calculated CRC, set e.result = > + * EC_RES_INVALID_HEADER_CRC and go to 301 > + * 103) If q.data_crc_present, calculate data CRC. If that mismatches the CRC > + * byte at the end of the packet, set e.result = EC_RES_INVALID_DATA_CRC > + * and go to 301. > + * 104) If q.seq_dup == 0, go to 201. > + * 105) If q.seq_num != r.seq_num, go to 201. > + * 106) If q.seq_dup == r.seq_dup, go to 205, else go to 203. > + * > + * 201) Process request q into response r. > + * 202) r.seq_num = q.seq_num > + * 203) r.seq_dup = q.seq_dup > + * 204) Calculate r.header_crc > + * 205) If r.data_len > 0 and data is no longer available, set e.result = > + * EC_RES_DUP_UNAVAILABLE and go to 301. > + * 206) Send response r. > + * > + * 301) e.seq_num = q.seq_num > + * 302) e.seq_dup = q.seq_dup > + * 303) Calculate e.header_crc. > + * 304) Send error response e. > + */ > + > +/* Version 4 request from host */ > +struct ec_host_request4 { > + /* > + * bits 0-3: struct_version: Structure version (=4) > + * bit 4: is_response: Is response (=0) > + * bits 5-6: seq_num: Sequence number > + * bit 7: seq_dup: Sequence duplicate flag > + */ > + uint8_t fields0; > + > + /* > + * bits 0-4: command_version: Command version > + * bits 5-6: Reserved (set 0, ignore on read) > + * bit 7: data_crc_present: Is data CRC present after data > + */ > + uint8_t fields1; > + > + /* Command code (EC_CMD_*) */ > + uint16_t command; > + > + /* Length of data which follows this header (not including data CRC) */ > + uint16_t data_len; > + > + /* Reserved (set 0, ignore on read) */ > + uint8_t reserved; > + > + /* CRC-8 of above fields, using x^8 + x^2 + x + 1 polynomial */ > + uint8_t header_crc; > +} __ec_align4; > + > +/* Version 4 response from EC */ > +struct ec_host_response4 { > + /* > + * bits 0-3: struct_version: Structure version (=4) > + * bit 4: is_response: Is response (=1) > + * bits 5-6: seq_num: Sequence number > + * bit 7: seq_dup: Sequence duplicate flag > + */ > + uint8_t fields0; > + > + /* > + * bits 0-6: Reserved (set 0, ignore on read) > + * bit 7: data_crc_present: Is data CRC present after data > + */ > + uint8_t fields1; > + > + /* Result code (EC_RES_*) */ > + uint16_t result; > + > + /* Length of data which follows this header (not including data CRC) */ > + uint16_t data_len; > + > + /* Reserved (set 0, ignore on read) */ > + uint8_t reserved; > + > + /* CRC-8 of above fields, using x^8 + x^2 + x + 1 polynomial */ > + uint8_t header_crc; > +} __ec_align4; > + > +/* Fields in fields0 byte */ > +#define EC_PACKET4_0_STRUCT_VERSION_MASK 0x0f > +#define EC_PACKET4_0_IS_RESPONSE_MASK 0x10 > +#define EC_PACKET4_0_SEQ_NUM_SHIFT 5 > +#define EC_PACKET4_0_SEQ_NUM_MASK 0x60 > +#define EC_PACKET4_0_SEQ_DUP_MASK 0x80 > + > +/* Fields in fields1 byte */ > +#define EC_PACKET4_1_COMMAND_VERSION_MASK 0x1f /* (request only) */ > +#define EC_PACKET4_1_DATA_CRC_PRESENT_MASK 0x80 > > /*****************************************************************************/ > /* > * Notes on commands: > * > * Each command is an 16-bit command value. Commands which take params or > - * return response data specify structs for that data. If no struct is > + * return response data specify structures for that data. If no structure is > * specified, the command does not input or output data, respectively. > * Parameter/response length is implicit in the structs. Some underlying > * communication protocols (I2C, SPI) may add length or checksum headers, but > * those are implementation-dependent and not defined here. > + * > + * All commands MUST be #defined to be 4-digit UPPER CASE hex values > + * (e.g., 0x00AB, not 0xab) for CONFIG_HOSTCMD_SECTION_SORTED to work. > */ > > /*****************************************************************************/ > @@ -530,7 +991,7 @@ struct ec_host_response { > * Get protocol version, used to deal with non-backward compatible protocol > * changes. > */ > -#define EC_CMD_PROTO_VERSION 0x00 > +#define EC_CMD_PROTO_VERSION 0x0000 > > /** > * struct ec_response_proto_version - Response to the proto version command. > @@ -538,13 +999,13 @@ struct ec_host_response { > */ > struct ec_response_proto_version { > uint32_t version; > -} __packed; > +} __ec_align4; > > /* > * Hello. This is a simple command to test the EC is responsive to > * commands. > */ > -#define EC_CMD_HELLO 0x01 > +#define EC_CMD_HELLO 0x0001 > > /** > * struct ec_params_hello - Parameters to the hello command. > @@ -552,7 +1013,7 @@ struct ec_response_proto_version { > */ > struct ec_params_hello { > uint32_t in_data; > -} __packed; > +} __ec_align4; > > /** > * struct ec_response_hello - Response to the hello command. > @@ -560,10 +1021,10 @@ struct ec_params_hello { > */ > struct ec_response_hello { > uint32_t out_data; > -} __packed; > +} __ec_align4; > > /* Get version number */ > -#define EC_CMD_GET_VERSION 0x02 > +#define EC_CMD_GET_VERSION 0x0002 > > enum ec_current_image { > EC_IMAGE_UNKNOWN = 0, > @@ -583,10 +1044,10 @@ struct ec_response_get_version { > char version_string_rw[32]; > char reserved[32]; > uint32_t current_image; > -} __packed; > +} __ec_align4; > > /* Read test */ > -#define EC_CMD_READ_TEST 0x03 > +#define EC_CMD_READ_TEST 0x0003 > > /** > * struct ec_params_read_test - Parameters for the read test command. > @@ -596,7 +1057,7 @@ struct ec_response_get_version { > struct ec_params_read_test { > uint32_t offset; > uint32_t size; > -} __packed; > +} __ec_align4; > > /** > * struct ec_response_read_test - Response to the read test command. > @@ -604,17 +1065,17 @@ struct ec_params_read_test { > */ > struct ec_response_read_test { > uint32_t data[32]; > -} __packed; > +} __ec_align4; > > /* > * Get build information > * > * Response is null-terminated string. > */ > -#define EC_CMD_GET_BUILD_INFO 0x04 > +#define EC_CMD_GET_BUILD_INFO 0x0004 > > /* Get chip info */ > -#define EC_CMD_GET_CHIP_INFO 0x05 > +#define EC_CMD_GET_CHIP_INFO 0x0005 > > /** > * struct ec_response_get_chip_info - Response to the get chip info command. > @@ -626,10 +1087,10 @@ struct ec_response_get_chip_info { > char vendor[32]; > char name[32]; > char revision[32]; > -} __packed; > +} __ec_align4; > > /* Get board HW version */ > -#define EC_CMD_GET_BOARD_VERSION 0x06 > +#define EC_CMD_GET_BOARD_VERSION 0x0006 > > /** > * struct ec_response_board_version - Response to the board version command. > @@ -637,7 +1098,7 @@ struct ec_response_get_chip_info { > */ > struct ec_response_board_version { > uint16_t board_version; > -} __packed; > +} __ec_align2; > > /* > * Read memory-mapped data. > @@ -647,7 +1108,7 @@ struct ec_response_board_version { > * > * Response is params.size bytes of data. > */ > -#define EC_CMD_READ_MEMMAP 0x07 > +#define EC_CMD_READ_MEMMAP 0x0007 > > /** > * struct ec_params_read_memmap - Parameters for the read memory map command. > @@ -657,10 +1118,10 @@ struct ec_response_board_version { > struct ec_params_read_memmap { > uint8_t offset; > uint8_t size; > -} __packed; > +} __ec_align1; > > /* Read versions supported for a command */ > -#define EC_CMD_GET_CMD_VERSIONS 0x08 > +#define EC_CMD_GET_CMD_VERSIONS 0x0008 > > /** > * struct ec_params_get_cmd_versions - Parameters for the get command versions. > @@ -668,7 +1129,7 @@ struct ec_params_read_memmap { > */ > struct ec_params_get_cmd_versions { > uint8_t cmd; > -} __packed; > +} __ec_align1; > > /** > * struct ec_params_get_cmd_versions_v1 - Parameters for the get command > @@ -677,7 +1138,7 @@ struct ec_params_get_cmd_versions { > */ > struct ec_params_get_cmd_versions_v1 { > uint16_t cmd; > -} __packed; > +} __ec_align2; > > /** > * struct ec_response_get_cmd_version - Response to the get command versions. > @@ -686,16 +1147,16 @@ struct ec_params_get_cmd_versions_v1 { > */ > struct ec_response_get_cmd_versions { > uint32_t version_mask; > -} __packed; > +} __ec_align4; > > /* > - * Check EC communcations status (busy). This is needed on i2c/spi but not > + * Check EC communications status (busy). This is needed on i2c/spi but not > * on lpc since it has its own out-of-band busy indicator. > * > * lpc must read the status from the command register. Attempting this on > * lpc will overwrite the args/parameter space and corrupt its data. > */ > -#define EC_CMD_GET_COMMS_STATUS 0x09 > +#define EC_CMD_GET_COMMS_STATUS 0x0009 > > /* Avoid using ec_status which is for return values */ > enum ec_comms_status { > @@ -709,25 +1170,25 @@ enum ec_comms_status { > */ > struct ec_response_get_comms_status { > uint32_t flags; /* Mask of enum ec_comms_status */ > -} __packed; > +} __ec_align4; > > /* Fake a variety of responses, purely for testing purposes. */ > -#define EC_CMD_TEST_PROTOCOL 0x0a > +#define EC_CMD_TEST_PROTOCOL 0x000A > > /* Tell the EC what to send back to us. */ > struct ec_params_test_protocol { > uint32_t ec_result; > uint32_t ret_len; > uint8_t buf[32]; > -} __packed; > +} __ec_align4; > > /* Here it comes... */ > struct ec_response_test_protocol { > uint8_t buf[32]; > -} __packed; > +} __ec_align4; > > -/* Get prococol information */ > -#define EC_CMD_GET_PROTOCOL_INFO 0x0b > +/* Get protocol information */ > +#define EC_CMD_GET_PROTOCOL_INFO 0x000B > > /* Flags for ec_response_get_protocol_info.flags */ > /* EC_RES_IN_PROGRESS may be returned if a command is slow */ > @@ -747,7 +1208,7 @@ struct ec_response_get_protocol_info { > uint16_t max_request_packet_size; > uint16_t max_response_packet_size; > uint32_t flags; > -} __packed; > +} __ec_align4; > > > /*****************************************************************************/ > @@ -765,19 +1226,19 @@ struct ec_response_get_protocol_info { > struct ec_params_get_set_value { > uint32_t flags; > uint32_t value; > -} __packed; > +} __ec_align4; > > struct ec_response_get_set_value { > uint32_t flags; > uint32_t value; > -} __packed; > +} __ec_align4; > > -/* More than one command can use these structs to get/set paramters. */ > -#define EC_CMD_GSV_PAUSE_IN_S5 0x0c > +/* More than one command can use these structs to get/set parameters. */ > +#define EC_CMD_GSV_PAUSE_IN_S5 0x000C > > /*****************************************************************************/ > /* List the features supported by the firmware */ > -#define EC_CMD_GET_FEATURES 0x0d > +#define EC_CMD_GET_FEATURES 0x000D > > /* Supported features */ > enum ec_feature_code { > @@ -840,7 +1301,7 @@ enum ec_feature_code { > * (Common Smart Battery System Interface Specification) > */ > EC_FEATURE_SMART_BATTERY = 18, > - /* EC can dectect when the host hangs. */ > + /* EC can detect when the host hangs. */ > EC_FEATURE_HANG_DETECT = 19, > /* Report power information, for pit only */ > EC_FEATURE_PMU = 20, > @@ -852,10 +1313,42 @@ enum ec_feature_code { > EC_FEATURE_USB_MUX = 23, > /* Motion Sensor code has an internal software FIFO */ > EC_FEATURE_MOTION_SENSE_FIFO = 24, > + /* Support temporary secure vstore */ > + EC_FEATURE_VSTORE = 25, > + /* EC decides on USB-C SS mux state, muxes configured by host */ > + EC_FEATURE_USBC_SS_MUX_VIRTUAL = 26, > /* EC has RTC feature that can be controlled by host commands */ > EC_FEATURE_RTC = 27, > + /* The MCU exposes a Fingerprint sensor */ > + EC_FEATURE_FINGERPRINT = 28, > + /* The MCU exposes a Touchpad */ > + EC_FEATURE_TOUCHPAD = 29, > + /* The MCU has RWSIG task enabled */ > + EC_FEATURE_RWSIG = 30, > + /* EC has device events support */ > + EC_FEATURE_DEVICE_EVENT = 31, > + /* EC supports the unified wake masks for LPC/eSPI systems */ > + EC_FEATURE_UNIFIED_WAKE_MASKS = 32, > + /* EC supports 64-bit host events */ > + EC_FEATURE_HOST_EVENT64 = 33, > + /* EC runs code in RAM (not in place, a.k.a. XIP) */ > + EC_FEATURE_EXEC_IN_RAM = 34, > /* EC supports CEC commands */ > EC_FEATURE_CEC = 35, > + /* EC supports tight sensor timestamping. */ > + EC_FEATURE_MOTION_SENSE_TIGHT_TIMESTAMPS = 36, > + /* > + * EC supports tablet mode detection aligned to Chrome and allows > + * setting of threshold by host command using > + * MOTIONSENSE_CMD_TABLET_MODE_LID_ANGLE. > + */ > + EC_FEATURE_REFINED_TABLET_MODE_HYSTERESIS = 37, > + /* EC supports audio codec. */ > + EC_FEATURE_AUDIO_CODEC = 38, > + /* EC Supports SCP. */ > + EC_FEATURE_SCP = 39, > + /* The MCU is Intel Integrated Sensor Hub */ > + EC_FEATURE_ISH = 40, > }; > > #define EC_FEATURE_MASK_0(event_code) (1UL << (event_code % 32)) > @@ -863,13 +1356,25 @@ enum ec_feature_code { > > struct ec_response_get_features { > uint32_t flags[2]; > -} __packed; > +} __ec_align4; > + > +/*****************************************************************************/ > +/* Get the board's SKU ID from EC */ > +#define EC_CMD_GET_SKU_ID 0x000E > + > +/* Set SKU ID from AP */ > +#define EC_CMD_SET_SKU_ID 0x000F > + > +struct ec_sku_id_info { > + uint32_t sku_id; > +} __ec_align4; > > /*****************************************************************************/ > /* Flash commands */ > > /* Get flash info */ > -#define EC_CMD_FLASH_INFO 0x10 > +#define EC_CMD_FLASH_INFO 0x0010 > +#define EC_VER_FLASH_INFO 2 > > /** > * struct ec_response_flash_info - Response to the flash info command. > @@ -888,12 +1393,23 @@ struct ec_response_flash_info { > uint32_t write_block_size; > uint32_t erase_block_size; > uint32_t protect_block_size; > -} __packed; > +} __ec_align4; > > -/* Flags for version 1+ flash info command */ > -/* EC flash erases bits to 0 instead of 1 */ > +/* > + * Flags for version 1+ flash info command > + * EC flash erases bits to 0 instead of 1. > + */ > #define EC_FLASH_INFO_ERASE_TO_0 (1 << 0) > > +/* > + * Flash must be selected for read/write/erase operations to succeed. This may > + * be necessary on a chip where write/erase can be corrupted by other board > + * activity, or where the chip needs to enable some sort of > programming voltage, > + * or where the read/write/erase operations require cleanly suspending other > + * chip functionality. > + */ > +#define EC_FLASH_INFO_SELECT_REQUIRED (1 << 1) > + > /** > * struct ec_response_flash_info_1 - Response to the flash info v1 command. > * @flash_size: Usable flash size in bytes. > @@ -914,7 +1430,14 @@ struct ec_response_flash_info { > * fields following. > * > * gcc anonymous structs don't seem to get along with the __packed directive; > - * if they did we'd define the version 0 struct as a sub-struct of this one. > + * if they did we'd define the version 0 structure as a sub-structure of this > + * one. > + * > + * Version 2 supports flash banks of different sizes: > + * The caller specified the number of banks it has preallocated > + * (num_banks_desc) > + * The EC returns the number of banks describing the flash memory. > + * It adds banks descriptions up to num_banks_desc. > */ > struct ec_response_flash_info_1 { > /* Version 0 fields; see above for description */ > @@ -926,14 +1449,50 @@ struct ec_response_flash_info_1 { > /* Version 1 adds these fields: */ > uint32_t write_ideal_size; > uint32_t flags; > -} __packed; > +} __ec_align4; > + > +struct ec_params_flash_info_2 { > + /* Number of banks to describe */ > + uint16_t num_banks_desc; > + /* Reserved; set 0; ignore on read */ > + uint8_t reserved[2]; > +} __ec_align4; > + > +struct ec_flash_bank { > + /* Number of sector is in this bank. */ > + uint16_t count; > + /* Size in power of 2 of each sector (8 --> 256 bytes) */ > + uint8_t size_exp; > + /* Minimal write size for the sectors in this bank */ > + uint8_t write_size_exp; > + /* Erase size for the sectors in this bank */ > + uint8_t erase_size_exp; > + /* Size for write protection, usually identical to erase size. */ > + uint8_t protect_size_exp; > + /* Reserved; set 0; ignore on read */ > + uint8_t reserved[2]; > +}; > + > +struct ec_response_flash_info_2 { > + /* Total flash in the EC. */ > + uint32_t flash_size; > + /* Flags; see EC_FLASH_INFO_* */ > + uint32_t flags; > + /* Maximum size to use to send data to write to the EC. */ > + uint32_t write_ideal_size; > + /* Number of banks present in the EC. */ > + uint16_t num_banks_total; > + /* Number of banks described in banks array. */ > + uint16_t num_banks_desc; > + struct ec_flash_bank banks[0]; > +} __ec_align4; > > /* > * Read flash > * > * Response is params.size bytes of data. > */ > -#define EC_CMD_FLASH_READ 0x11 > +#define EC_CMD_FLASH_READ 0x0011 > > /** > * struct ec_params_flash_read - Parameters for the flash read command. > @@ -943,10 +1502,10 @@ struct ec_response_flash_info_1 { > struct ec_params_flash_read { > uint32_t offset; > uint32_t size; > -} __packed; > +} __ec_align4; > > /* Write flash */ > -#define EC_CMD_FLASH_WRITE 0x12 > +#define EC_CMD_FLASH_WRITE 0x0012 > #define EC_VER_FLASH_WRITE 1 > > /* Version 0 of the flash command supported only 64 bytes of data */ > @@ -961,32 +1520,69 @@ struct ec_params_flash_write { > uint32_t offset; > uint32_t size; > /* Followed by data to write */ > -} __packed; > +} __ec_align4; > > /* Erase flash */ > -#define EC_CMD_FLASH_ERASE 0x13 > +#define EC_CMD_FLASH_ERASE 0x0013 > > /** > - * struct ec_params_flash_erase - Parameters for the flash erase command. > + * struct ec_params_flash_erase - Parameters for the flash erase command, v0. > * @offset: Byte offset to erase. > * @size: Size to erase in bytes. > */ > struct ec_params_flash_erase { > uint32_t offset; > uint32_t size; > -} __packed; > +} __ec_align4; > > /* > - * Get/set flash protection. > + * v1 add async erase: > + * subcommands can returns: > + * EC_RES_SUCCESS : erased (see ERASE_SECTOR_ASYNC case below). > + * EC_RES_INVALID_PARAM : offset/size are not aligned on a erase boundary. > + * EC_RES_ERROR : other errors. > + * EC_RES_BUSY : an existing erase operation is in progress. > + * EC_RES_ACCESS_DENIED: Trying to erase running image. > * > - * If mask!=0, sets/clear the requested bits of flags. Depending on the > - * firmware write protect GPIO, not all flags will take effect immediately; > - * some flags require a subsequent hard reset to take effect. Check the > + * When ERASE_SECTOR_ASYNC returns EC_RES_SUCCESS, the operation is just > + * properly queued. The user must call ERASE_GET_RESULT subcommand to get > + * the proper result. > + * When ERASE_GET_RESULT returns EC_RES_BUSY, the caller must wait and send > + * ERASE_GET_RESULT again to get the result of ERASE_SECTOR_ASYNC. > + * ERASE_GET_RESULT command may timeout on EC where flash access is not > + * permitted while erasing. (For instance, STM32F4). > + */ > +enum ec_flash_erase_cmd { > + FLASH_ERASE_SECTOR, /* Erase and wait for result */ > + FLASH_ERASE_SECTOR_ASYNC, /* Erase and return immediately. */ > + FLASH_ERASE_GET_RESULT, /* Ask for last erase result */ > +}; > + > +/** > + * struct ec_params_flash_erase_v1 - Parameters for the flash erase > command, v1. > + * @cmd: One of ec_flash_erase_cmd. > + * @reserved: Pad byte; currently always contains 0. > + * @flag: No flags defined yet; set to 0. > + * @params: Same as v0 parameters. > + */ > +struct ec_params_flash_erase_v1 { > + uint8_t cmd; > + uint8_t reserved; > + uint16_t flag; > + struct ec_params_flash_erase params; > +} __ec_align4; > + > +/* > + * Get/set flash protection. > + * > + * If mask!=0, sets/clear the requested bits of flags. Depending on the > + * firmware write protect GPIO, not all flags will take effect immediately; > + * some flags require a subsequent hard reset to take effect. Check the > * returned flags bits to see what actually happened. > * > * If mask=0, simply returns the current flags state. > */ > -#define EC_CMD_FLASH_PROTECT 0x15 > +#define EC_CMD_FLASH_PROTECT 0x0015 > #define EC_VER_FLASH_PROTECT 1 /* Command version 1 */ > > /* Flags for flash protection */ > @@ -1009,8 +1605,17 @@ struct ec_params_flash_erase { > * re-requesting the desired flags, or by a hard reset if that fails. > */ > #define EC_FLASH_PROTECT_ERROR_INCONSISTENT (1 << 5) > -/* Entile flash code protected when the EC boots */ > +/* Entire flash code protected when the EC boots */ > #define EC_FLASH_PROTECT_ALL_AT_BOOT (1 << 6) > +/* RW flash code protected when the EC boots */ > +#define EC_FLASH_PROTECT_RW_AT_BOOT (1 << 7) > +/* RW flash code protected now. */ > +#define EC_FLASH_PROTECT_RW_NOW (1 << 8) > +/* Rollback information flash region protected when the EC boots */ > +#define EC_FLASH_PROTECT_ROLLBACK_AT_BOOT (1 << 9) > +/* Rollback information flash region protected now */ > +#define EC_FLASH_PROTECT_ROLLBACK_NOW (1 << 10) > + > > /** > * struct ec_params_flash_protect - Parameters for the flash protect command. > @@ -1020,7 +1625,7 @@ struct ec_params_flash_erase { > struct ec_params_flash_protect { > uint32_t mask; > uint32_t flags; > -} __packed; > +} __ec_align4; > > /** > * struct ec_response_flash_protect - Response to the flash protect command. > @@ -1035,7 +1640,7 @@ struct ec_response_flash_protect { > uint32_t flags; > uint32_t valid_flags; > uint32_t writable_flags; > -} __packed; > +} __ec_align4; > > /* > * Note: commands 0x14 - 0x19 version 0 were old commands to get/set flash > @@ -1043,22 +1648,37 @@ struct ec_response_flash_protect { > */ > > /* Get the region offset/size */ > -#define EC_CMD_FLASH_REGION_INFO 0x16 > +#define EC_CMD_FLASH_REGION_INFO 0x0016 > #define EC_VER_FLASH_REGION_INFO 1 > > enum ec_flash_region { > /* Region which holds read-only EC image */ > EC_FLASH_REGION_RO = 0, > - /* Region which holds rewritable EC image */ > - EC_FLASH_REGION_RW, > + /* > + * Region which holds active RW image. 'Active' is different from > + * 'running'. Active means 'scheduled-to-run'. Since RO image always > + * scheduled to run, active/non-active applies only to RW images (for > + * the same reason 'update' applies only to RW images. It's a state of > + * an image on a flash. Running image can be RO, RW_A, RW_B but active > + * image can only be RW_A or RW_B. In recovery mode, an active RW image > + * doesn't enter 'running' state but it's still active on a flash. > + */ > + EC_FLASH_REGION_ACTIVE, > /* > * Region which should be write-protected in the factory (a superset of > * EC_FLASH_REGION_RO) > */ > EC_FLASH_REGION_WP_RO, > + /* Region which holds updatable (non-active) RW image */ > + EC_FLASH_REGION_UPDATE, > /* Number of regions */ > EC_FLASH_REGION_COUNT, > }; > +/* > + * 'RW' is vague if there are multiple RW images; we mean the active one, > + * so the old constant is deprecated. > + */ > +#define EC_FLASH_REGION_RW EC_FLASH_REGION_ACTIVE > > /** > * struct ec_params_flash_region_info - Parameters for the flash region info > @@ -1067,15 +1687,15 @@ enum ec_flash_region { > */ > struct ec_params_flash_region_info { > uint32_t region; > -} __packed; > +} __ec_align4; > > struct ec_response_flash_region_info { > uint32_t offset; > uint32_t size; > -} __packed; > +} __ec_align4; > > /* Read/write VbNvContext */ > -#define EC_CMD_VBNV_CONTEXT 0x17 > +#define EC_CMD_VBNV_CONTEXT 0x0017 > #define EC_VER_VBNV_CONTEXT 1 > #define EC_VBNV_BLOCK_SIZE 16 > > @@ -1087,52 +1707,99 @@ enum ec_vbnvcontext_op { > struct ec_params_vbnvcontext { > uint32_t op; > uint8_t block[EC_VBNV_BLOCK_SIZE]; > -} __packed; > +} __ec_align4; > > struct ec_response_vbnvcontext { > uint8_t block[EC_VBNV_BLOCK_SIZE]; > -} __packed; > +} __ec_align4; > + > + > +/* Get SPI flash information */ > +#define EC_CMD_FLASH_SPI_INFO 0x0018 > + > +struct ec_response_flash_spi_info { > + /* JEDEC info from command 0x9F (manufacturer, memory type, size) */ > + uint8_t jedec[3]; > + > + /* Pad byte; currently always contains 0 */ > + uint8_t reserved0; > + > + /* Manufacturer / device ID from command 0x90 */ > + uint8_t mfr_dev_id[2]; > + > + /* Status registers from command 0x05 and 0x35 */ > + uint8_t sr1, sr2; > +} __ec_align1; > + > + > +/* Select flash during flash operations */ > +#define EC_CMD_FLASH_SELECT 0x0019 > + > +/** > + * struct ec_params_flash_select - Parameters for the flash select command. > + * @select: 1 to select flash, 0 to deselect flash > + */ > +struct ec_params_flash_select { > + uint8_t select; > +} __ec_align4; > + > > /*****************************************************************************/ > /* PWM commands */ > > /* Get fan target RPM */ > -#define EC_CMD_PWM_GET_FAN_TARGET_RPM 0x20 > +#define EC_CMD_PWM_GET_FAN_TARGET_RPM 0x0020 > > struct ec_response_pwm_get_fan_rpm { > uint32_t rpm; > -} __packed; > +} __ec_align4; > > /* Set target fan RPM */ > -#define EC_CMD_PWM_SET_FAN_TARGET_RPM 0x21 > +#define EC_CMD_PWM_SET_FAN_TARGET_RPM 0x0021 > + > +/* Version 0 of input params */ > +struct ec_params_pwm_set_fan_target_rpm_v0 { > + uint32_t rpm; > +} __ec_align4; > > -struct ec_params_pwm_set_fan_target_rpm { > +/* Version 1 of input params */ > +struct ec_params_pwm_set_fan_target_rpm_v1 { > uint32_t rpm; > -} __packed; > + uint8_t fan_idx; > +} __ec_align_size1; > > /* Get keyboard backlight */ > -#define EC_CMD_PWM_GET_KEYBOARD_BACKLIGHT 0x22 > +/* OBSOLETE - Use EC_CMD_PWM_SET_DUTY */ > +#define EC_CMD_PWM_GET_KEYBOARD_BACKLIGHT 0x0022 > > struct ec_response_pwm_get_keyboard_backlight { > uint8_t percent; > uint8_t enabled; > -} __packed; > +} __ec_align1; > > /* Set keyboard backlight */ > -#define EC_CMD_PWM_SET_KEYBOARD_BACKLIGHT 0x23 > +/* OBSOLETE - Use EC_CMD_PWM_SET_DUTY */ > +#define EC_CMD_PWM_SET_KEYBOARD_BACKLIGHT 0x0023 > > struct ec_params_pwm_set_keyboard_backlight { > uint8_t percent; > -} __packed; > +} __ec_align1; > > /* Set target fan PWM duty cycle */ > -#define EC_CMD_PWM_SET_FAN_DUTY 0x24 > +#define EC_CMD_PWM_SET_FAN_DUTY 0x0024 > + > +/* Version 0 of input params */ > +struct ec_params_pwm_set_fan_duty_v0 { > + uint32_t percent; > +} __ec_align4; > > -struct ec_params_pwm_set_fan_duty { > +/* Version 1 of input params */ > +struct ec_params_pwm_set_fan_duty_v1 { > uint32_t percent; > -} __packed; > + uint8_t fan_idx; > +} __ec_align_size1; > > -#define EC_CMD_PWM_SET_DUTY 0x25 > +#define EC_CMD_PWM_SET_DUTY 0x0025 > /* 16 bit duty cycle, 0xffff = 100% */ > #define EC_PWM_MAX_DUTY 0xffff > > @@ -1150,18 +1817,18 @@ struct ec_params_pwm_set_duty { > uint16_t duty; /* Duty cycle, EC_PWM_MAX_DUTY = 100% */ > uint8_t pwm_type; /* ec_pwm_type */ > uint8_t index; /* Type-specific index, or 0 if unique */ > -} __packed; > +} __ec_align4; > > -#define EC_CMD_PWM_GET_DUTY 0x26 > +#define EC_CMD_PWM_GET_DUTY 0x0026 > > struct ec_params_pwm_get_duty { > uint8_t pwm_type; /* ec_pwm_type */ > uint8_t index; /* Type-specific index, or 0 if unique */ > -} __packed; > +} __ec_align1; > > struct ec_response_pwm_get_duty { > uint16_t duty; /* Duty cycle, EC_PWM_MAX_DUTY = 100% */ > -} __packed; > +} __ec_align2; > > /*****************************************************************************/ > /* > @@ -1170,11 +1837,11 @@ struct ec_response_pwm_get_duty { > * into a subcommand. We'll make separate structs for subcommands with > * different input args, so that we know how much to expect. > */ > -#define EC_CMD_LIGHTBAR_CMD 0x28 > +#define EC_CMD_LIGHTBAR_CMD 0x0028 > > struct rgb_s { > uint8_t r, g, b; > -}; > +} __ec_todo_unpacked; > > #define LB_BATTERY_LEVELS 4 > > @@ -1214,7 +1881,7 @@ struct lightbar_params_v0 { > > /* Color palette */ > struct rgb_s color[8]; /* 0-3 are Google colors */ > -} __packed; > +} __ec_todo_packed; > > struct lightbar_params_v1 { > /* Timing */ > @@ -1227,9 +1894,72 @@ struct lightbar_params_v1 { > int32_t s3_sleep_for; > int32_t s3_ramp_up; > int32_t s3_ramp_down; > + int32_t s5_ramp_up; > + int32_t s5_ramp_down; > + int32_t tap_tick_delay; > + int32_t tap_gate_delay; > + int32_t tap_display_time; > + > + /* Tap-for-battery params */ > + uint8_t tap_pct_red; > + uint8_t tap_pct_green; > + uint8_t tap_seg_min_on; > + uint8_t tap_seg_max_on; > + uint8_t tap_seg_osc; > + uint8_t tap_idx[3]; > + > + /* Oscillation */ > + uint8_t osc_min[2]; /* AC=0/1 */ > + uint8_t osc_max[2]; /* AC=0/1 */ > + uint8_t w_ofs[2]; /* AC=0/1 */ > + > + /* Brightness limits based on the backlight and AC. */ > + uint8_t bright_bl_off_fixed[2]; /* AC=0/1 */ > + uint8_t bright_bl_on_min[2]; /* AC=0/1 */ > + uint8_t bright_bl_on_max[2]; /* AC=0/1 */ > + > + /* Battery level thresholds */ > + uint8_t battery_threshold[LB_BATTERY_LEVELS - 1]; > + > + /* Map [AC][battery_level] to color index */ > + uint8_t s0_idx[2][LB_BATTERY_LEVELS]; /* AP is running */ > + uint8_t s3_idx[2][LB_BATTERY_LEVELS]; /* AP is sleeping */ > + > + /* s5: single color pulse on inhibited power-up */ > + uint8_t s5_idx; > + > + /* Color palette */ > + struct rgb_s color[8]; /* 0-3 are Google colors */ > +} __ec_todo_packed; > + > +/* Lightbar command params v2 > + * crbug.com/467716 > + * > + * lightbar_parms_v1 was too big for i2c, therefore in v2, we split them up by > + * logical groups to make it more manageable ( < 120 bytes). > + * > + * NOTE: Each of these groups must be less than 120 bytes. > + */ > + > +struct lightbar_params_v2_timing { > + /* Timing */ > + int32_t google_ramp_up; > + int32_t google_ramp_down; > + int32_t s3s0_ramp_up; > + int32_t s0_tick_delay[2]; /* AC=0/1 */ > + int32_t s0a_tick_delay[2]; /* AC=0/1 */ > + int32_t s0s3_ramp_down; > + int32_t s3_sleep_for; > + int32_t s3_ramp_up; > + int32_t s3_ramp_down; > + int32_t s5_ramp_up; > + int32_t s5_ramp_down; > int32_t tap_tick_delay; > + int32_t tap_gate_delay; > int32_t tap_display_time; > +} __ec_todo_packed; > > +struct lightbar_params_v2_tap { > /* Tap-for-battery params */ > uint8_t tap_pct_red; > uint8_t tap_pct_green; > @@ -1237,102 +1967,140 @@ struct lightbar_params_v1 { > uint8_t tap_seg_max_on; > uint8_t tap_seg_osc; > uint8_t tap_idx[3]; > +} __ec_todo_packed; > > +struct lightbar_params_v2_oscillation { > /* Oscillation */ > uint8_t osc_min[2]; /* AC=0/1 */ > uint8_t osc_max[2]; /* AC=0/1 */ > uint8_t w_ofs[2]; /* AC=0/1 */ > +} __ec_todo_packed; > > +struct lightbar_params_v2_brightness { > /* Brightness limits based on the backlight and AC. */ > uint8_t bright_bl_off_fixed[2]; /* AC=0/1 */ > uint8_t bright_bl_on_min[2]; /* AC=0/1 */ > uint8_t bright_bl_on_max[2]; /* AC=0/1 */ > +} __ec_todo_packed; > > +struct lightbar_params_v2_thresholds { > /* Battery level thresholds */ > uint8_t battery_threshold[LB_BATTERY_LEVELS - 1]; > +} __ec_todo_packed; > > +struct lightbar_params_v2_colors { > /* Map [AC][battery_level] to color index */ > uint8_t s0_idx[2][LB_BATTERY_LEVELS]; /* AP is running */ > uint8_t s3_idx[2][LB_BATTERY_LEVELS]; /* AP is sleeping */ > > + /* s5: single color pulse on inhibited power-up */ > + uint8_t s5_idx; > + > /* Color palette */ > struct rgb_s color[8]; /* 0-3 are Google colors */ > -} __packed; > +} __ec_todo_packed; > > -/* Lightbar program */ > +/* Lightbyte program. */ > #define EC_LB_PROG_LEN 192 > struct lightbar_program { > uint8_t size; > uint8_t data[EC_LB_PROG_LEN]; > -}; > +} __ec_todo_unpacked; > > struct ec_params_lightbar { > uint8_t cmd; /* Command (see enum lightbar_command) */ > union { > - struct { > - /* no args */ > - } dump, off, on, init, get_seq, get_params_v0, get_params_v1, > - version, get_brightness, get_demo, suspend, resume; > + /* > + * The following commands have no args: > + * > + * dump, off, on, init, get_seq, get_params_v0, get_params_v1, > + * version, get_brightness, get_demo, suspend, resume, > + * get_params_v2_timing, get_params_v2_tap, get_params_v2_osc, > + * get_params_v2_bright, get_params_v2_thlds, > + * get_params_v2_colors > + * > + * Don't use an empty struct, because C++ hates that. > + */ > > - struct { > + struct __ec_todo_unpacked { > uint8_t num; > } set_brightness, seq, demo; > > - struct { > + struct __ec_todo_unpacked { > uint8_t ctrl, reg, value; > } reg; > > - struct { > + struct __ec_todo_unpacked { > uint8_t led, red, green, blue; > } set_rgb; > > - struct { > + struct __ec_todo_unpacked { > uint8_t led; > } get_rgb; > > - struct { > + struct __ec_todo_unpacked { > uint8_t enable; > } manual_suspend_ctrl; > > struct lightbar_params_v0 set_params_v0; > struct lightbar_params_v1 set_params_v1; > + > + struct lightbar_params_v2_timing set_v2par_timing; > + struct lightbar_params_v2_tap set_v2par_tap; > + struct lightbar_params_v2_oscillation set_v2par_osc; > + struct lightbar_params_v2_brightness set_v2par_bright; > + struct lightbar_params_v2_thresholds set_v2par_thlds; > + struct lightbar_params_v2_colors set_v2par_colors; > + > struct lightbar_program set_program; > }; > -} __packed; > +} __ec_todo_packed; > > struct ec_response_lightbar { > union { > - struct { > - struct { > + struct __ec_todo_unpacked { > + struct __ec_todo_unpacked { > uint8_t reg; > uint8_t ic0; > uint8_t ic1; > } vals[23]; > } dump; > > - struct { > + struct __ec_todo_unpacked { > uint8_t num; > } get_seq, get_brightness, get_demo; > > struct lightbar_params_v0 get_params_v0; > struct lightbar_params_v1 get_params_v1; > > - struct { > + > + struct lightbar_params_v2_timing get_params_v2_timing; > + struct lightbar_params_v2_tap get_params_v2_tap; > + struct lightbar_params_v2_oscillation get_params_v2_osc; > + struct lightbar_params_v2_brightness get_params_v2_bright; > + struct lightbar_params_v2_thresholds get_params_v2_thlds; > + struct lightbar_params_v2_colors get_params_v2_colors; > + > + struct __ec_todo_unpacked { > uint32_t num; > uint32_t flags; > } version; > > - struct { > + struct __ec_todo_unpacked { > uint8_t red, green, blue; > } get_rgb; > > - struct { > - /* no return params */ > - } off, on, init, set_brightness, seq, reg, set_rgb, > - demo, set_params_v0, set_params_v1, > - set_program, manual_suspend_ctrl, suspend, resume; > + /* > + * The following commands have no response: > + * > + * off, on, init, set_brightness, seq, reg, set_rgb, demo, > + * set_params_v0, set_params_v1, set_program, > + * manual_suspend_ctrl, suspend, resume, set_v2par_timing, > + * set_v2par_tap, set_v2par_osc, set_v2par_bright, > + * set_v2par_thlds, set_v2par_colors > + */ > }; > -} __packed; > +} __ec_todo_packed; > > /* Lightbar commands */ > enum lightbar_command { > @@ -1358,13 +2126,25 @@ enum lightbar_command { > LIGHTBAR_CMD_MANUAL_SUSPEND_CTRL = 19, > LIGHTBAR_CMD_SUSPEND = 20, > LIGHTBAR_CMD_RESUME = 21, > + LIGHTBAR_CMD_GET_PARAMS_V2_TIMING = 22, > + LIGHTBAR_CMD_SET_PARAMS_V2_TIMING = 23, > + LIGHTBAR_CMD_GET_PARAMS_V2_TAP = 24, > + LIGHTBAR_CMD_SET_PARAMS_V2_TAP = 25, > + LIGHTBAR_CMD_GET_PARAMS_V2_OSCILLATION = 26, > + LIGHTBAR_CMD_SET_PARAMS_V2_OSCILLATION = 27, > + LIGHTBAR_CMD_GET_PARAMS_V2_BRIGHTNESS = 28, > + LIGHTBAR_CMD_SET_PARAMS_V2_BRIGHTNESS = 29, > + LIGHTBAR_CMD_GET_PARAMS_V2_THRESHOLDS = 30, > + LIGHTBAR_CMD_SET_PARAMS_V2_THRESHOLDS = 31, > + LIGHTBAR_CMD_GET_PARAMS_V2_COLORS = 32, > + LIGHTBAR_CMD_SET_PARAMS_V2_COLORS = 33, > LIGHTBAR_NUM_CMDS > }; > > /*****************************************************************************/ > /* LED control commands */ > > -#define EC_CMD_LED_CONTROL 0x29 > +#define EC_CMD_LED_CONTROL 0x0029 > > enum ec_led_id { > /* LED to indicate battery state of charge */ > @@ -1376,6 +2156,14 @@ enum ec_led_id { > EC_LED_ID_POWER_LED, > /* LED on power adapter or its plug */ > EC_LED_ID_ADAPTER_LED, > + /* LED to indicate left side */ > + EC_LED_ID_LEFT_LED, > + /* LED to indicate right side */ > + EC_LED_ID_RIGHT_LED, > + /* LED to indicate recovery mode with HW_REINIT */ > + EC_LED_ID_RECOVERY_HW_REINIT_LED, > + /* LED to indicate sysrq debug mode. */ > + EC_LED_ID_SYSRQ_DEBUG_LED, > > EC_LED_ID_COUNT > }; > @@ -1390,6 +2178,7 @@ enum ec_led_colors { > EC_LED_COLOR_BLUE, > EC_LED_COLOR_YELLOW, > EC_LED_COLOR_WHITE, > + EC_LED_COLOR_AMBER, > > EC_LED_COLOR_COUNT > }; > @@ -1399,7 +2188,7 @@ struct ec_params_led_control { > uint8_t flags; /* Control flags */ > > uint8_t brightness[EC_LED_COLOR_COUNT]; > -} __packed; > +} __ec_align1; > > struct ec_response_led_control { > /* > @@ -1410,7 +2199,7 @@ struct ec_response_led_control { > * Other values means the LED is control by PWM. > */ > uint8_t brightness_range[EC_LED_COLOR_COUNT]; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* Verified boot commands */ > @@ -1421,7 +2210,7 @@ struct ec_response_led_control { > */ > > /* Verified boot hash command */ > -#define EC_CMD_VBOOT_HASH 0x2A > +#define EC_CMD_VBOOT_HASH 0x002A > > struct ec_params_vboot_hash { > uint8_t cmd; /* enum ec_vboot_hash_cmd */ > @@ -1431,7 +2220,7 @@ struct ec_params_vboot_hash { > uint32_t offset; /* Offset in flash to hash */ > uint32_t size; /* Number of bytes to hash */ > uint8_t nonce_data[64]; /* Nonce data; ignored if nonce_size=0 */ > -} __packed; > +} __ec_align4; > > struct ec_response_vboot_hash { > uint8_t status; /* enum ec_vboot_hash_status */ > @@ -1441,7 +2230,7 @@ struct ec_response_vboot_hash { > uint32_t offset; /* Offset in flash which was hashed */ > uint32_t size; /* Number of bytes hashed */ > uint8_t hash_digest[64]; /* Hash digest data */ > -} __packed; > +} __ec_align4; > > enum ec_vboot_hash_cmd { > EC_VBOOT_HASH_GET = 0, /* Get current hash status */ > @@ -1465,15 +2254,22 @@ enum ec_vboot_hash_status { > * If one of these is specified, the EC will automatically update offset and > * size to the correct values for the specified image (RO or RW). > */ > -#define EC_VBOOT_HASH_OFFSET_RO 0xfffffffe > -#define EC_VBOOT_HASH_OFFSET_RW 0xfffffffd > +#define EC_VBOOT_HASH_OFFSET_RO 0xfffffffe > +#define EC_VBOOT_HASH_OFFSET_ACTIVE 0xfffffffd > +#define EC_VBOOT_HASH_OFFSET_UPDATE 0xfffffffc > + > +/* > + * 'RW' is vague if there are multiple RW images; we mean the active one, > + * so the old constant is deprecated. > + */ > +#define EC_VBOOT_HASH_OFFSET_RW EC_VBOOT_HASH_OFFSET_ACTIVE > > /*****************************************************************************/ > /* > * Motion sense commands. We'll make separate structs for sub-commands with > * different input args, so that we know how much to expect. > */ > -#define EC_CMD_MOTION_SENSE_CMD 0x2B > +#define EC_CMD_MOTION_SENSE_CMD 0x002B > > /* Motion sense commands */ > enum motionsense_command { > @@ -1492,7 +2288,13 @@ enum motionsense_command { > > /* > * EC Rate command is a setter/getter command for the EC sampling rate > - * of all motion sensors in milliseconds. > + * in milliseconds. > + * It is per sensor, the EC run sample task at the minimum of all > + * sensors EC_RATE. > + * For sensors without hardware FIFO, EC_RATE should be equals to 1/ODR > + * to collect all the sensor samples. > + * For sensor with hardware FIFO, EC_RATE is used as the maximal delay > + * to process of all motion sensors in milliseconds. > */ > MOTIONSENSE_CMD_EC_RATE = 2, > > @@ -1523,32 +2325,76 @@ enum motionsense_command { > MOTIONSENSE_CMD_DATA = 6, > > /* > - * Perform low level calibration.. On sensors that support it, ask to > - * do offset calibration. > + * Return sensor fifo info. > + */ > + MOTIONSENSE_CMD_FIFO_INFO = 7, > + > + /* > + * Insert a flush element in the fifo and return sensor fifo info. > + * The host can use that element to synchronize its operation. > + */ > + MOTIONSENSE_CMD_FIFO_FLUSH = 8, > + > + /* > + * Return a portion of the fifo. > + */ > + MOTIONSENSE_CMD_FIFO_READ = 9, > + > + /* > + * Perform low level calibration. > + * On sensors that support it, ask to do offset calibration. > */ > MOTIONSENSE_CMD_PERFORM_CALIB = 10, > > /* > - * Sensor Offset command is a setter/getter command for the offset used > - * for calibration. The offsets can be calculated by the host, or via > + * Sensor Offset command is a setter/getter command for the offset > + * used for calibration. > + * The offsets can be calculated by the host, or via > * PERFORM_CALIB command. > */ > MOTIONSENSE_CMD_SENSOR_OFFSET = 11, > > - /* Number of motionsense sub-commands. */ > - MOTIONSENSE_NUM_CMDS > -}; > + /* > + * List available activities for a MOTION sensor. > + * Indicates if they are enabled or disabled. > + */ > + MOTIONSENSE_CMD_LIST_ACTIVITIES = 12, > + > + /* > + * Activity management > + * Enable/Disable activity recognition. > + */ > + MOTIONSENSE_CMD_SET_ACTIVITY = 13, > + > + /* > + * Lid Angle > + */ > + MOTIONSENSE_CMD_LID_ANGLE = 14, > + > + /* > + * Allow the FIFO to trigger interrupt via MKBP events. > + * By default the FIFO does not send interrupt to process the FIFO > + * until the AP is ready or it is coming from a wakeup sensor. > + */ > + MOTIONSENSE_CMD_FIFO_INT_ENABLE = 15, > > -enum motionsensor_id { > - EC_MOTION_SENSOR_ACCEL_BASE = 0, > - EC_MOTION_SENSOR_ACCEL_LID = 1, > - EC_MOTION_SENSOR_GYRO = 2, > + /* > + * Spoof the readings of the sensors. The spoofed readings can be set > + * to arbitrary values, or will lock to the last read actual values. > + */ > + MOTIONSENSE_CMD_SPOOF = 16, > + > + /* Set lid angle for tablet mode detection. */ > + MOTIONSENSE_CMD_TABLET_MODE_LID_ANGLE = 17, > > /* > - * Note, if more sensors are added and this count changes, the padding > - * in ec_response_motion_sense dump command must be modified. > + * Sensor Scale command is a setter/getter command for the calibration > + * scale. > */ > - EC_MOTION_SENSOR_COUNT = 3 > + MOTIONSENSE_CMD_SENSOR_SCALE = 18, > + > + /* Number of motionsense sub-commands. */ > + MOTIONSENSE_NUM_CMDS > }; > > /* List of motion sensor types. */ > @@ -1560,6 +2406,7 @@ enum motionsensor_type { > MOTIONSENSE_TYPE_LIGHT = 4, > MOTIONSENSE_TYPE_ACTIVITY = 5, > MOTIONSENSE_TYPE_BARO = 6, > + MOTIONSENSE_TYPE_SYNC = 7, > MOTIONSENSE_TYPE_MAX, > }; > > @@ -1567,20 +2414,114 @@ enum motionsensor_type { > enum motionsensor_location { > MOTIONSENSE_LOC_BASE = 0, > MOTIONSENSE_LOC_LID = 1, > + MOTIONSENSE_LOC_CAMERA = 2, > MOTIONSENSE_LOC_MAX, > }; > > /* List of motion sensor chips. */ > enum motionsensor_chip { > MOTIONSENSE_CHIP_KXCJ9 = 0, > + MOTIONSENSE_CHIP_LSM6DS0 = 1, > + MOTIONSENSE_CHIP_BMI160 = 2, > + MOTIONSENSE_CHIP_SI1141 = 3, > + MOTIONSENSE_CHIP_SI1142 = 4, > + MOTIONSENSE_CHIP_SI1143 = 5, > + MOTIONSENSE_CHIP_KX022 = 6, > + MOTIONSENSE_CHIP_L3GD20H = 7, > + MOTIONSENSE_CHIP_BMA255 = 8, > + MOTIONSENSE_CHIP_BMP280 = 9, > + MOTIONSENSE_CHIP_OPT3001 = 10, > + MOTIONSENSE_CHIP_BH1730 = 11, > + MOTIONSENSE_CHIP_GPIO = 12, > + MOTIONSENSE_CHIP_LIS2DH = 13, > + MOTIONSENSE_CHIP_LSM6DSM = 14, > + MOTIONSENSE_CHIP_LIS2DE = 15, > + MOTIONSENSE_CHIP_LIS2MDL = 16, > + MOTIONSENSE_CHIP_LSM6DS3 = 17, > + MOTIONSENSE_CHIP_MAX, > +}; > + > +/* List of orientation positions */ > +enum motionsensor_orientation { > + MOTIONSENSE_ORIENTATION_LANDSCAPE = 0, > + MOTIONSENSE_ORIENTATION_PORTRAIT = 1, > + MOTIONSENSE_ORIENTATION_UPSIDE_DOWN_PORTRAIT = 2, > + MOTIONSENSE_ORIENTATION_UPSIDE_DOWN_LANDSCAPE = 3, > + MOTIONSENSE_ORIENTATION_UNKNOWN = 4, > +}; > + > +struct ec_response_motion_sensor_data { > + /* Flags for each sensor. */ > + uint8_t flags; > + /* sensor number the data comes from */ > + uint8_t sensor_num; > + /* Each sensor is up to 3-axis. */ > + union { > + int16_t data[3]; > + struct __ec_todo_packed { > + uint16_t reserved; > + uint32_t timestamp; > + }; > + struct __ec_todo_unpacked { > + uint8_t activity; /* motionsensor_activity */ > + uint8_t state; > + int16_t add_info[2]; > + }; > + }; > +} __ec_todo_packed; > + > +/* Note: used in ec_response_get_next_data */ > +struct ec_response_motion_sense_fifo_info { > + /* Size of the fifo */ > + uint16_t size; > + /* Amount of space used in the fifo */ > + uint16_t count; > + /* Timestamp recorded in us. > + * aka accurate timestamp when host event was triggered. > + */ > + uint32_t timestamp; > + /* Total amount of vector lost */ > + uint16_t total_lost; > + /* Lost events since the last fifo_info, per sensors */ > + uint16_t lost[0]; > +} __ec_todo_packed; > + > +struct ec_response_motion_sense_fifo_data { > + uint32_t number_data; > + struct ec_response_motion_sensor_data data[0]; > +} __ec_todo_packed; > + > +/* List supported activity recognition */ > +enum motionsensor_activity { > + MOTIONSENSE_ACTIVITY_RESERVED = 0, > + MOTIONSENSE_ACTIVITY_SIG_MOTION = 1, > + MOTIONSENSE_ACTIVITY_DOUBLE_TAP = 2, > + MOTIONSENSE_ACTIVITY_ORIENTATION = 3, > }; > > +struct ec_motion_sense_activity { > + uint8_t sensor_num; > + uint8_t activity; /* one of enum motionsensor_activity */ > + uint8_t enable; /* 1: enable, 0: disable */ > + uint8_t reserved; > + uint16_t parameters[3]; /* activity dependent parameters */ > +} __ec_todo_unpacked; > + > /* Module flag masks used for the dump sub-command. */ > #define MOTIONSENSE_MODULE_FLAG_ACTIVE (1<<0) > > /* Sensor flag masks used for the dump sub-command. */ > #define MOTIONSENSE_SENSOR_FLAG_PRESENT (1<<0) > > +/* > + * Flush entry for synchronization. > + * data contains time stamp > + */ > +#define MOTIONSENSE_SENSOR_FLAG_FLUSH (1<<0) > +#define MOTIONSENSE_SENSOR_FLAG_TIMESTAMP (1<<1) > +#define MOTIONSENSE_SENSOR_FLAG_WAKEUP (1<<2) > +#define MOTIONSENSE_SENSOR_FLAG_TABLET_MODE (1<<3) > + > /* > * Send this value for the data element to only perform a read. If you > * send any other value, the EC will interpret it as data to set and will > @@ -1590,48 +2531,79 @@ enum motionsensor_chip { > > #define EC_MOTION_SENSE_INVALID_CALIB_TEMP 0x8000 > > +/* MOTIONSENSE_CMD_SENSOR_OFFSET subcommand flag */ > /* Set Calibration information */ > -#define MOTION_SENSE_SET_OFFSET 1 > +#define MOTION_SENSE_SET_OFFSET (1 << 0) > > -struct ec_response_motion_sensor_data { > - /* Flags for each sensor. */ > - uint8_t flags; > - /* Sensor number the data comes from */ > - uint8_t sensor_num; > - /* Each sensor is up to 3-axis. */ > - union { > - int16_t data[3]; > - struct { > - uint16_t rsvd; > - uint32_t timestamp; > - } __packed; > - struct { > - uint8_t activity; /* motionsensor_activity */ > - uint8_t state; > - int16_t add_info[2]; > - }; > - }; > -} __packed; > +/* Default Scale value, factor 1. */ > +#define MOTION_SENSE_DEFAULT_SCALE (1 << 15) > + > +#define LID_ANGLE_UNRELIABLE 500 > + > +enum motionsense_spoof_mode { > + /* Disable spoof mode. */ > + MOTIONSENSE_SPOOF_MODE_DISABLE = 0, > + > + /* Enable spoof mode, but use provided component values. */ > + MOTIONSENSE_SPOOF_MODE_CUSTOM, > + > + /* Enable spoof mode, but use the current sensor values. */ > + MOTIONSENSE_SPOOF_MODE_LOCK_CURRENT, > + > + /* Query the current spoof mode status for the sensor. */ > + MOTIONSENSE_SPOOF_MODE_QUERY, > +}; > > struct ec_params_motion_sense { > uint8_t cmd; > union { > - /* Used for MOTIONSENSE_CMD_DUMP. */ > - struct { > - /* no args */ > + /* Used for MOTIONSENSE_CMD_DUMP */ > + struct __ec_todo_unpacked { > + /* > + * Maximal number of sensor the host is expecting. > + * 0 means the host is only interested in the number > + * of sensors controlled by the EC. > + */ > + uint8_t max_sensor_count; > } dump; > > /* > - * Used for MOTIONSENSE_CMD_EC_RATE and > - * MOTIONSENSE_CMD_KB_WAKE_ANGLE. > + * Used for MOTIONSENSE_CMD_KB_WAKE_ANGLE. > */ > - struct { > - /* Data to set or EC_MOTION_SENSE_NO_VALUE to read. */ > + struct __ec_todo_unpacked { > + /* Data to set or EC_MOTION_SENSE_NO_VALUE to read. > + * kb_wake_angle: angle to wakup AP. > + */ > int16_t data; > - } ec_rate, kb_wake_angle; > + } kb_wake_angle; > + > + /* > + * Used for MOTIONSENSE_CMD_INFO, MOTIONSENSE_CMD_DATA > + * and MOTIONSENSE_CMD_PERFORM_CALIB. > + */ > + struct __ec_todo_unpacked { > + uint8_t sensor_num; > + } info, info_3, data, fifo_flush, perform_calib, > + list_activities; > + > + /* > + * Used for MOTIONSENSE_CMD_EC_RATE, MOTIONSENSE_CMD_SENSOR_ODR > + * and MOTIONSENSE_CMD_SENSOR_RANGE. > + */ > + struct __ec_todo_unpacked { > + uint8_t sensor_num; > + > + /* Rounding flag, true for round-up, false for down. */ > + uint8_t roundup; > + > + uint16_t reserved; > + > + /* Data to set or EC_MOTION_SENSE_NO_VALUE to read. */ > + int32_t data; > + } ec_rate, sensor_odr, sensor_range; > > /* Used for MOTIONSENSE_CMD_SENSOR_OFFSET */ > - struct { > + struct __ec_todo_packed { > uint8_t sensor_num; > > /* > @@ -1657,51 +2629,117 @@ struct ec_params_motion_sense { > * Compass: 1/16 uT > */ > int16_t offset[3]; > - } __packed sensor_offset; > + } sensor_offset; > > - /* Used for MOTIONSENSE_CMD_INFO. */ > - struct { > + /* Used for MOTIONSENSE_CMD_SENSOR_SCALE */ > + struct __ec_todo_packed { > uint8_t sensor_num; > - } info; > > - /* > - * Used for MOTIONSENSE_CMD_SENSOR_ODR and > - * MOTIONSENSE_CMD_SENSOR_RANGE. > - */ > - struct { > - /* Should be element of enum motionsensor_id. */ > - uint8_t sensor_num; > + /* > + * bit 0: If set (MOTION_SENSE_SET_OFFSET), set > + * the calibration information in the EC. > + * If unset, just retrieve calibration information. > + */ > + uint16_t flags; > > - /* Rounding flag, true for round-up, false for down. */ > - uint8_t roundup; > + /* > + * Temperature at calibration, in units of 0.01 C > + * 0x8000: invalid / unknown. > + * 0x0: 0C > + * 0x7fff: +327.67C > + */ > + int16_t temp; > > - uint16_t reserved; > + /* > + * Scale for calibration: > + * By default scale is 1, it is encoded on 16bits: > + * 1 = 1 << 15 > + * ~2 = 0xFFFF > + * ~0 = 0. > + */ > + uint16_t scale[3]; > + } sensor_scale; > > - /* Data to set or EC_MOTION_SENSE_NO_VALUE to read. */ > - int32_t data; > - } sensor_odr, sensor_range; > + > + /* Used for MOTIONSENSE_CMD_FIFO_INFO */ > + /* (no params) */ > + > + /* Used for MOTIONSENSE_CMD_FIFO_READ */ > + struct __ec_todo_unpacked { > + /* > + * Number of expected vector to return. > + * EC may return less or 0 if none available. > + */ > + uint32_t max_data_vector; > + } fifo_read; > + > + struct ec_motion_sense_activity set_activity; > + > + /* Used for MOTIONSENSE_CMD_LID_ANGLE */ > + /* (no params) */ > + > + /* Used for MOTIONSENSE_CMD_FIFO_INT_ENABLE */ > + struct __ec_todo_unpacked { > + /* > + * 1: enable, 0 disable fifo, > + * EC_MOTION_SENSE_NO_VALUE return value. > + */ > + int8_t enable; > + } fifo_int_enable; > + > + /* Used for MOTIONSENSE_CMD_SPOOF */ > + struct __ec_todo_packed { > + uint8_t sensor_id; > + > + /* See enum motionsense_spoof_mode. */ > + uint8_t spoof_enable; > + > + /* Ignored, used for alignment. */ > + uint8_t reserved; > + > + /* Individual component values to spoof. */ > + int16_t components[3]; > + } spoof; > + > + /* Used for MOTIONSENSE_CMD_TABLET_MODE_LID_ANGLE. */ > + struct __ec_todo_unpacked { > + /* > + * Lid angle threshold for switching between tablet and > + * clamshell mode. > + */ > + int16_t lid_angle; > + > + /* > + * Hysteresis degree to prevent fluctuations between > + * clamshell and tablet mode if lid angle keeps > + * changing around the threshold. Lid motion driver will > + * use lid_angle + hys_degree to trigger tablet mode and > + * lid_angle - hys_degree to trigger clamshell mode. > + */ > + int16_t hys_degree; > + } tablet_mode_threshold; > }; > -} __packed; > +} __ec_todo_packed; > > struct ec_response_motion_sense { > union { > - /* Used for MOTIONSENSE_CMD_DUMP. */ > - struct { > + /* Used for MOTIONSENSE_CMD_DUMP */ > + struct __ec_todo_unpacked { > /* Flags representing the motion sensor module. */ > uint8_t module_flags; > > - /* Number of sensors managed directly by the EC. */ > + /* Number of sensors managed directly by the EC */ > uint8_t sensor_count; > > /* > - * Sensor data is truncated if response_max is too small > + * sensor data is truncated if response_max is too small > * for holding all the data. > */ > struct ec_response_motion_sensor_data sensor[0]; > } dump; > > /* Used for MOTIONSENSE_CMD_INFO. */ > - struct { > + struct __ec_todo_unpacked { > /* Should be element of enum motionsensor_type. */ > uint8_t type; > > @@ -1712,74 +2750,166 @@ struct ec_response_motion_sense { > uint8_t chip; > } info; > > + /* Used for MOTIONSENSE_CMD_INFO version 3 */ > + struct __ec_todo_unpacked { > + /* Should be element of enum motionsensor_type. */ > + uint8_t type; > + > + /* Should be element of enum motionsensor_location. */ > + uint8_t location; > + > + /* Should be element of enum motionsensor_chip. */ > + uint8_t chip; > + > + /* Minimum sensor sampling frequency */ > + uint32_t min_frequency; > + > + /* Maximum sensor sampling frequency */ > + uint32_t max_frequency; > + > + /* Max number of sensor events that could be in fifo */ > + uint32_t fifo_max_event_count; > + } info_3; > + > /* Used for MOTIONSENSE_CMD_DATA */ > struct ec_response_motion_sensor_data data; > > /* > * Used for MOTIONSENSE_CMD_EC_RATE, MOTIONSENSE_CMD_SENSOR_ODR, > - * MOTIONSENSE_CMD_SENSOR_RANGE, and > - * MOTIONSENSE_CMD_KB_WAKE_ANGLE. > + * MOTIONSENSE_CMD_SENSOR_RANGE, > + * MOTIONSENSE_CMD_KB_WAKE_ANGLE, > + * MOTIONSENSE_CMD_FIFO_INT_ENABLE and > + * MOTIONSENSE_CMD_SPOOF. > */ > - struct { > + struct __ec_todo_unpacked { > /* Current value of the parameter queried. */ > int32_t ret; > - } ec_rate, sensor_odr, sensor_range, kb_wake_angle; > + } ec_rate, sensor_odr, sensor_range, kb_wake_angle, > + fifo_int_enable, spoof; > > - /* Used for MOTIONSENSE_CMD_SENSOR_OFFSET */ > - struct { > + /* > + * Used for MOTIONSENSE_CMD_SENSOR_OFFSET, > + * PERFORM_CALIB. > + */ > + struct __ec_todo_unpacked { > int16_t temp; > int16_t offset[3]; > } sensor_offset, perform_calib; > - }; > -} __packed; > > -/*****************************************************************************/ > -/* USB charging control commands */ > + /* Used for MOTIONSENSE_CMD_SENSOR_SCALE */ > + struct __ec_todo_unpacked { > + int16_t temp; > + uint16_t scale[3]; > + } sensor_scale; > > -/* Set USB port charging mode */ > -#define EC_CMD_USB_CHARGE_SET_MODE 0x30 > + struct ec_response_motion_sense_fifo_info fifo_info, fifo_flush; > > -struct ec_params_usb_charge_set_mode { > - uint8_t usb_port_id; > - uint8_t mode; > -} __packed; > + struct ec_response_motion_sense_fifo_data fifo_read; > > -/*****************************************************************************/ > -/* Persistent storage for host */ > + struct __ec_todo_packed { > + uint16_t reserved; > + uint32_t enabled; > + uint32_t disabled; > + } list_activities; > + > + /* No params for set activity */ > + > + /* Used for MOTIONSENSE_CMD_LID_ANGLE */ > + struct __ec_todo_unpacked { > + /* > + * Angle between 0 and 360 degree if available, > + * LID_ANGLE_UNRELIABLE otherwise. > + */ > + uint16_t value; > + } lid_angle; > + > + /* Used for MOTIONSENSE_CMD_TABLET_MODE_LID_ANGLE. */ > + struct __ec_todo_unpacked { > + /* > + * Lid angle threshold for switching between tablet and > + * clamshell mode. > + */ > + uint16_t lid_angle; > + > + /* Hysteresis degree. */ > + uint16_t hys_degree; > + } tablet_mode_threshold; > + > + }; > +} __ec_todo_packed; > + > +/*****************************************************************************/ > +/* Force lid open command */ > + > +/* Make lid event always open */ > +#define EC_CMD_FORCE_LID_OPEN 0x002C > + > +struct ec_params_force_lid_open { > + uint8_t enabled; > +} __ec_align1; > + > +/*****************************************************************************/ > +/* Configure the behavior of the power button */ > +#define EC_CMD_CONFIG_POWER_BUTTON 0x002D > + > +enum ec_config_power_button_flags { > + /* Enable/Disable power button pulses for x86 devices */ > + EC_POWER_BUTTON_ENABLE_PULSE = (1 << 0), > +}; > + > +struct ec_params_config_power_button { > + /* See enum ec_config_power_button_flags */ > + uint8_t flags; > +} __ec_align1; > + > +/*****************************************************************************/ > +/* USB charging control commands */ > + > +/* Set USB port charging mode */ > +#define EC_CMD_USB_CHARGE_SET_MODE 0x0030 > + > +struct ec_params_usb_charge_set_mode { > + uint8_t usb_port_id; > + uint8_t mode:7; > + uint8_t inhibit_charge:1; > +} __ec_align1; > + > +/*****************************************************************************/ > +/* Persistent storage for host */ > > /* Maximum bytes that can be read/written in a single command */ > #define EC_PSTORE_SIZE_MAX 64 > > /* Get persistent storage info */ > -#define EC_CMD_PSTORE_INFO 0x40 > +#define EC_CMD_PSTORE_INFO 0x0040 > > struct ec_response_pstore_info { > /* Persistent storage size, in bytes */ > uint32_t pstore_size; > /* Access size; read/write offset and size must be a multiple of this */ > uint32_t access_size; > -} __packed; > +} __ec_align4; > > /* > * Read persistent storage > * > * Response is params.size bytes of data. > */ > -#define EC_CMD_PSTORE_READ 0x41 > +#define EC_CMD_PSTORE_READ 0x0041 > > struct ec_params_pstore_read { > uint32_t offset; /* Byte offset to read */ > uint32_t size; /* Size to read in bytes */ > -} __packed; > +} __ec_align4; > > /* Write persistent storage */ > -#define EC_CMD_PSTORE_WRITE 0x42 > +#define EC_CMD_PSTORE_WRITE 0x0042 > > struct ec_params_pstore_write { > uint32_t offset; /* Byte offset to write */ > uint32_t size; /* Size to write in bytes */ > uint8_t data[EC_PSTORE_SIZE_MAX]; > -} __packed; > +} __ec_align4; > > /*****************************************************************************/ > /* Real-time clock */ > @@ -1787,21 +2917,21 @@ struct ec_params_pstore_write { > /* RTC params and response structures */ > struct ec_params_rtc { > uint32_t time; > -} __packed; > +} __ec_align4; > > struct ec_response_rtc { > uint32_t time; > -} __packed; > +} __ec_align4; > > /* These use ec_response_rtc */ > -#define EC_CMD_RTC_GET_VALUE 0x44 > -#define EC_CMD_RTC_GET_ALARM 0x45 > +#define EC_CMD_RTC_GET_VALUE 0x0044 > +#define EC_CMD_RTC_GET_ALARM 0x0045 > > /* These all use ec_params_rtc */ > -#define EC_CMD_RTC_SET_VALUE 0x46 > -#define EC_CMD_RTC_SET_ALARM 0x47 > +#define EC_CMD_RTC_SET_VALUE 0x0046 > +#define EC_CMD_RTC_SET_ALARM 0x0047 > > -/* Pass as param to SET_ALARM to clear the current alarm */ > +/* Pass as time param to SET_ALARM to clear the current alarm */ > #define EC_RTC_ALARM_CLEAR 0 > > /*****************************************************************************/ > @@ -1811,8 +2941,8 @@ struct ec_response_rtc { > #define EC_PORT80_SIZE_MAX 32 > > /* Get last port80 code from previous boot */ > -#define EC_CMD_PORT80_LAST_BOOT 0x48 > -#define EC_CMD_PORT80_READ 0x48 > +#define EC_CMD_PORT80_LAST_BOOT 0x0048 > +#define EC_CMD_PORT80_READ 0x0048 > > enum ec_port80_subcmd { > EC_PORT80_GET_INFO = 0, > @@ -1822,29 +2952,72 @@ enum ec_port80_subcmd { > struct ec_params_port80_read { > uint16_t subcmd; > union { > - struct { > + struct __ec_todo_unpacked { > uint32_t offset; > uint32_t num_entries; > } read_buffer; > }; > -} __packed; > +} __ec_todo_packed; > > struct ec_response_port80_read { > union { > - struct { > + struct __ec_todo_unpacked { > uint32_t writes; > uint32_t history_size; > uint32_t last_boot; > } get_info; > - struct { > + struct __ec_todo_unpacked { > uint16_t codes[EC_PORT80_SIZE_MAX]; > } data; > }; > -} __packed; > +} __ec_todo_packed; > > struct ec_response_port80_last_boot { > uint16_t code; > -} __packed; > +} __ec_align2; > + > +/*****************************************************************************/ > +/* Temporary secure storage for host verified boot use */ > + > +/* Number of bytes in a vstore slot */ > +#define EC_VSTORE_SLOT_SIZE 64 > + > +/* Maximum number of vstore slots */ > +#define EC_VSTORE_SLOT_MAX 32 > + > +/* Get persistent storage info */ > +#define EC_CMD_VSTORE_INFO 0x0049 > +struct ec_response_vstore_info { > + /* Indicates which slots are locked */ > + uint32_t slot_locked; > + /* Total number of slots available */ > + uint8_t slot_count; > +} __ec_align_size1; > + > +/* > + * Read temporary secure storage > + * > + * Response is EC_VSTORE_SLOT_SIZE bytes of data. > + */ > +#define EC_CMD_VSTORE_READ 0x004A > + > +struct ec_params_vstore_read { > + uint8_t slot; /* Slot to read from */ > +} __ec_align1; > + > +struct ec_response_vstore_read { > + uint8_t data[EC_VSTORE_SLOT_SIZE]; > +} __ec_align1; > + > +/* > + * Write temporary secure storage and lock it. > + */ > +#define EC_CMD_VSTORE_WRITE 0x004B > + > +struct ec_params_vstore_write { > + uint8_t slot; /* Slot to write to */ > + uint8_t data[EC_VSTORE_SLOT_SIZE]; > +} __ec_align1; > > /*****************************************************************************/ > /* Thermal engine commands. Note that there are two implementations. We'll > @@ -1853,8 +3026,8 @@ struct ec_response_port80_last_boot { > * Version 1 separates the CPU thermal limits from the fan control. > */ > > -#define EC_CMD_THERMAL_SET_THRESHOLD 0x50 > -#define EC_CMD_THERMAL_GET_THRESHOLD 0x51 > +#define EC_CMD_THERMAL_SET_THRESHOLD 0x0050 > +#define EC_CMD_THERMAL_GET_THRESHOLD 0x0051 > > /* The version 0 structs are opaque. You have to know what they are for > * the get/set commands to make any sense. > @@ -1865,17 +3038,17 @@ struct ec_params_thermal_set_threshold { > uint8_t sensor_type; > uint8_t threshold_id; > uint16_t value; > -} __packed; > +} __ec_align2; > > /* Version 0 - get */ > struct ec_params_thermal_get_threshold { > uint8_t sensor_type; > uint8_t threshold_id; > -} __packed; > +} __ec_align1; > > struct ec_response_thermal_get_threshold { > uint16_t value; > -} __packed; > +} __ec_align2; > > > /* The version 1 structs are visible. */ > @@ -1887,19 +3060,39 @@ enum ec_temp_thresholds { > EC_TEMP_THRESH_COUNT > }; > > -/* Thermal configuration for one temperature sensor. Temps are in degrees K. > +/* > + * Thermal configuration for one temperature sensor. Temps are in degrees K. > * Zero values will be silently ignored by the thermal task. > + * > + * Set 'temp_host' value allows thermal task to trigger some event > with 1 degree > + * hysteresis. > + * For example, > + * temp_host[EC_TEMP_THRESH_HIGH] = 300 K > + * temp_host_release[EC_TEMP_THRESH_HIGH] = 0 K > + * EC will throttle ap when temperature >= 301 K, and release throttling when > + * temperature <= 299 K. > + * > + * Set 'temp_host_release' value allows thermal task has a custom hysteresis. > + * For example, > + * temp_host[EC_TEMP_THRESH_HIGH] = 300 K > + * temp_host_release[EC_TEMP_THRESH_HIGH] = 295 K > + * EC will throttle ap when temperature >= 301 K, and release throttling when > + * temperature <= 294 K. > + * > + * Note that this structure is a sub-structure of > + * ec_params_thermal_set_threshold_v1, but maintains its alignment there. > */ > struct ec_thermal_config { > uint32_t temp_host[EC_TEMP_THRESH_COUNT]; /* levels of hotness */ > + uint32_t temp_host_release[EC_TEMP_THRESH_COUNT]; /* release levels */ > uint32_t temp_fan_off; /* no active cooling needed */ > uint32_t temp_fan_max; /* max active cooling needed */ > -} __packed; > +} __ec_align4; > > /* Version 1 - get config for one sensor. */ > struct ec_params_thermal_get_threshold_v1 { > uint32_t sensor_num; > -} __packed; > +} __ec_align4; > /* This returns a struct ec_thermal_config */ > > /* Version 1 - set config for one sensor. > @@ -1907,51 +3100,82 @@ struct ec_params_thermal_get_threshold_v1 { > struct ec_params_thermal_set_threshold_v1 { > uint32_t sensor_num; > struct ec_thermal_config cfg; > -} __packed; > +} __ec_align4; > /* This returns no data */ > > /****************************************************************************/ > > /* Toggle automatic fan control */ > -#define EC_CMD_THERMAL_AUTO_FAN_CTRL 0x52 > +#define EC_CMD_THERMAL_AUTO_FAN_CTRL 0x0052 > > -/* Get TMP006 calibration data */ > -#define EC_CMD_TMP006_GET_CALIBRATION 0x53 > +/* Version 1 of input params */ > +struct ec_params_auto_fan_ctrl_v1 { > + uint8_t fan_idx; > +} __ec_align1; > + > +/* Get/Set TMP006 calibration data */ > +#define EC_CMD_TMP006_GET_CALIBRATION 0x0053 > +#define EC_CMD_TMP006_SET_CALIBRATION 0x0054 > + > +/* > + * The original TMP006 calibration only needed four params, but now we need > + * more. Since the algorithm is nothing but magic numbers anyway, we'll leave > + * the params opaque. The v1 "get" response will include the algorithm number > + * and how many params it requires. That way we can change the EC code without > + * needing to update this file. We can also use a different algorithm on each > + * sensor. > + */ > > +/* This is the same struct for both v0 and v1. */ > struct ec_params_tmp006_get_calibration { > uint8_t index; > -} __packed; > +} __ec_align1; > > -struct ec_response_tmp006_get_calibration { > +/* Version 0 */ > +struct ec_response_tmp006_get_calibration_v0 { > float s0; > float b0; > float b1; > float b2; > -} __packed; > +} __ec_align4; > > -/* Set TMP006 calibration data */ > -#define EC_CMD_TMP006_SET_CALIBRATION 0x54 > - > -struct ec_params_tmp006_set_calibration { > +struct ec_params_tmp006_set_calibration_v0 { > uint8_t index; > - uint8_t reserved[3]; /* Reserved; set 0 */ > + uint8_t reserved[3]; > float s0; > float b0; > float b1; > float b2; > -} __packed; > +} __ec_align4; > + > +/* Version 1 */ > +struct ec_response_tmp006_get_calibration_v1 { > + uint8_t algorithm; > + uint8_t num_params; > + uint8_t reserved[2]; > + float val[0]; > +} __ec_align4; > + > +struct ec_params_tmp006_set_calibration_v1 { > + uint8_t index; > + uint8_t algorithm; > + uint8_t num_params; > + uint8_t reserved; > + float val[0]; > +} __ec_align4; > + > > /* Read raw TMP006 data */ > -#define EC_CMD_TMP006_GET_RAW 0x55 > +#define EC_CMD_TMP006_GET_RAW 0x0055 > > struct ec_params_tmp006_get_raw { > uint8_t index; > -} __packed; > +} __ec_align1; > > struct ec_response_tmp006_get_raw { > int32_t t; /* In 1/100 K */ > int32_t v; /* In nV */ > -}; > +} __ec_align4; > > /*****************************************************************************/ > /* MKBP - Matrix KeyBoard Protocol */ > @@ -1966,24 +3190,24 @@ struct ec_response_tmp006_get_raw { > * to obtain the instantaneous state, use EC_CMD_MKBP_INFO with the type > * EC_MKBP_INFO_CURRENT and event EC_MKBP_EVENT_KEY_MATRIX. > */ > -#define EC_CMD_MKBP_STATE 0x60 > +#define EC_CMD_MKBP_STATE 0x0060 > > /* > * Provide information about various MKBP things. See enum ec_mkbp_info_type. > */ > -#define EC_CMD_MKBP_INFO 0x61 > +#define EC_CMD_MKBP_INFO 0x0061 > > struct ec_response_mkbp_info { > uint32_t rows; > uint32_t cols; > /* Formerly "switches", which was 0. */ > uint8_t reserved; > -} __packed; > +} __ec_align_size1; > > struct ec_params_mkbp_info { > uint8_t info_type; > uint8_t event_type; > -} __packed; > +} __ec_align1; > > enum ec_mkbp_info_type { > /* > @@ -2025,17 +3249,28 @@ enum ec_mkbp_info_type { > }; > > /* Simulate key press */ > -#define EC_CMD_MKBP_SIMULATE_KEY 0x62 > +#define EC_CMD_MKBP_SIMULATE_KEY 0x0062 > > struct ec_params_mkbp_simulate_key { > uint8_t col; > uint8_t row; > uint8_t pressed; > -} __packed; > +} __ec_align1; > + > +#define EC_CMD_GET_KEYBOARD_ID 0x0063 > + > +struct ec_response_keyboard_id { > + uint32_t keyboard_id; > +} __ec_align4; > + > +enum keyboard_id { > + KEYBOARD_ID_UNSUPPORTED = 0, > + KEYBOARD_ID_UNREADABLE = 0xffffffff, > +}; > > /* Configure keyboard scanning */ > -#define EC_CMD_MKBP_SET_CONFIG 0x64 > -#define EC_CMD_MKBP_GET_CONFIG 0x65 > +#define EC_CMD_MKBP_SET_CONFIG 0x0064 > +#define EC_CMD_MKBP_GET_CONFIG 0x0065 > > /* flags */ > enum mkbp_config_flags { > @@ -2052,7 +3287,12 @@ enum mkbp_config_valid { > EC_MKBP_VALID_FIFO_MAX_DEPTH = 1 << 7, > }; > > -/* Configuration for our key scanning algorithm */ > +/* > + * Configuration for our key scanning algorithm. > + * > + * Note that this is used as a sub-structure of > + * ec_{params/response}_mkbp_get_config. > + */ > struct ec_mkbp_config { > uint32_t valid_mask; /* valid fields */ > uint8_t flags; /* some flags (enum mkbp_config_flags) */ > @@ -2072,18 +3312,18 @@ struct ec_mkbp_config { > uint16_t debounce_up_us; /* time for debounce on key up */ > /* maximum depth to allow for fifo (0 = no keyscan output) */ > uint8_t fifo_max_depth; > -} __packed; > +} __ec_align_size1; > > struct ec_params_mkbp_set_config { > struct ec_mkbp_config config; > -} __packed; > +} __ec_align_size1; > > struct ec_response_mkbp_get_config { > struct ec_mkbp_config config; > -} __packed; > +} __ec_align_size1; > > /* Run the key scan emulation */ > -#define EC_CMD_KEYSCAN_SEQ_CTRL 0x66 > +#define EC_CMD_KEYSCAN_SEQ_CTRL 0x0066 > > enum ec_keyscan_seq_cmd { > EC_KEYSCAN_SEQ_STATUS = 0, /* Get status information */ > @@ -2103,18 +3343,18 @@ enum ec_collect_flags { > > struct ec_collect_item { > uint8_t flags; /* some flags (enum ec_collect_flags) */ > -}; > +} __ec_align1; > > struct ec_params_keyscan_seq_ctrl { > uint8_t cmd; /* Command to send (enum ec_keyscan_seq_cmd) */ > union { > - struct { > + struct __ec_align1 { > uint8_t active; /* still active */ > uint8_t num_items; /* number of items */ > /* Current item being presented */ > uint8_t cur_item; > } status; > - struct { > + struct __ec_todo_unpacked { > /* > * Absolute time for this scan, measured from the > * start of the sequence. > @@ -2122,29 +3362,40 @@ struct ec_params_keyscan_seq_ctrl { > uint32_t time_us; > uint8_t scan[0]; /* keyscan data */ > } add; > - struct { > + struct __ec_align1 { > uint8_t start_item; /* First item to return */ > uint8_t num_items; /* Number of items to return */ > } collect; > }; > -} __packed; > +} __ec_todo_packed; > > struct ec_result_keyscan_seq_ctrl { > union { > - struct { > + struct __ec_todo_unpacked { > uint8_t num_items; /* Number of items */ > /* Data for each item */ > struct ec_collect_item item[0]; > } collect; > }; > -} __packed; > +} __ec_todo_packed; > > /* > - * Command for retrieving the next pending MKBP event from the EC device > + * Get the next pending MKBP event. > * > - * The device replies with UNAVAILABLE if there aren't any pending events. > + * Returns EC_RES_UNAVAILABLE if there is no event pending. > + */ > +#define EC_CMD_GET_NEXT_EVENT 0x0067 > + > +#define EC_MKBP_HAS_MORE_EVENTS_SHIFT 7 > + > +/* > + * We use the most significant bit of the event type to indicate to the host > + * that the EC has more MKBP events available to provide. > */ > -#define EC_CMD_GET_NEXT_EVENT 0x67 > +#define EC_MKBP_HAS_MORE_EVENTS (1 << EC_MKBP_HAS_MORE_EVENTS_SHIFT) > + > +/* The mask to apply to get the raw event type */ > +#define EC_MKBP_EVENT_TYPE_MASK ((1 << EC_MKBP_HAS_MORE_EVENTS_SHIFT) - 1) > > enum ec_mkbp_event { > /* Keyboard matrix changed. The event data is the new matrix state. */ > @@ -2162,9 +3413,21 @@ enum ec_mkbp_event { > /* The state of the switches have changed. */ > EC_MKBP_EVENT_SWITCH = 4, > > - /* EC sent a sysrq command */ > + /* New Fingerprint sensor event, the event data is fp_events bitmap. */ > + EC_MKBP_EVENT_FINGERPRINT = 5, > + > + /* > + * Sysrq event: send emulated sysrq. The event data is sysrq, > + * corresponding to the key to be pressed. > + */ > EC_MKBP_EVENT_SYSRQ = 6, > > + /* > + * New 64-bit host event. > + * The event data is 8 bytes of host event flags. > + */ > + EC_MKBP_EVENT_HOST_EVENT64 = 7, > + > /* Notify the AP that something happened on CEC */ > EC_MKBP_EVENT_CEC_EVENT = 8, > > @@ -2174,65 +3437,140 @@ enum ec_mkbp_event { > /* Number of MKBP events */ > EC_MKBP_EVENT_COUNT, > }; > +BUILD_ASSERT(EC_MKBP_EVENT_COUNT <= EC_MKBP_EVENT_TYPE_MASK); > > -union ec_response_get_next_data { > - uint8_t key_matrix[13]; > +union __ec_align_offset1 ec_response_get_next_data { > + uint8_t key_matrix[13]; > > /* Unaligned */ > - uint32_t host_event; > + uint32_t host_event; > + uint64_t host_event64; > + > + struct __ec_todo_unpacked { > + /* For aligning the fifo_info */ > + uint8_t reserved[3]; > + struct ec_response_motion_sense_fifo_info info; > + } sensor_fifo; > + > + uint32_t buttons; > + > + uint32_t switches; > > - uint32_t buttons; > - uint32_t switches; > - uint32_t sysrq; > -} __packed; > + uint32_t fp_events; > > -union ec_response_get_next_data_v1 { > + uint32_t sysrq; > + > + /* CEC events from enum mkbp_cec_event */ > + uint32_t cec_events; > +}; > + > +union __ec_align_offset1 ec_response_get_next_data_v1 { > uint8_t key_matrix[16]; > + > + /* Unaligned */ > uint32_t host_event; > + uint64_t host_event64; > + > + struct __ec_todo_unpacked { > + /* For aligning the fifo_info */ > + uint8_t reserved[3]; > + struct ec_response_motion_sense_fifo_info info; > + } sensor_fifo; > + > uint32_t buttons; > + > uint32_t switches; > + > + uint32_t fp_events; > + > uint32_t sysrq; > + > + /* CEC events from enum mkbp_cec_event */ > uint32_t cec_events; > + > uint8_t cec_message[16]; > -} __packed; > +}; > +BUILD_ASSERT(sizeof(union ec_response_get_next_data_v1) == 16); > > struct ec_response_get_next_event { > uint8_t event_type; > /* Followed by event data if any */ > union ec_response_get_next_data data; > -} __packed; > +} __ec_align1; > > struct ec_response_get_next_event_v1 { > uint8_t event_type; > /* Followed by event data if any */ > union ec_response_get_next_data_v1 data; > -} __packed; > +} __ec_align1; > > /* Bit indices for buttons and switches.*/ > /* Buttons */ > #define EC_MKBP_POWER_BUTTON 0 > #define EC_MKBP_VOL_UP 1 > #define EC_MKBP_VOL_DOWN 2 > +#define EC_MKBP_RECOVERY 3 > > /* Switches */ > #define EC_MKBP_LID_OPEN 0 > #define EC_MKBP_TABLET_MODE 1 > #define EC_MKBP_BASE_ATTACHED 2 > > +/* Run keyboard factory test scanning */ > +#define EC_CMD_KEYBOARD_FACTORY_TEST 0x0068 > + > +struct ec_response_keyboard_factory_test { > + uint16_t shorted; /* Keyboard pins are shorted */ > +} __ec_align2; > + > +/* Fingerprint events in 'fp_events' for EC_MKBP_EVENT_FINGERPRINT */ > +#define EC_MKBP_FP_RAW_EVENT(fp_events) ((fp_events) & 0x00FFFFFF) > +#define EC_MKBP_FP_ERRCODE(fp_events) ((fp_events) & 0x0000000F) > +#define EC_MKBP_FP_ENROLL_PROGRESS_OFFSET 4 > +#define EC_MKBP_FP_ENROLL_PROGRESS(fpe) (((fpe) & 0x00000FF0) \ > + >> EC_MKBP_FP_ENROLL_PROGRESS_OFFSET) > +#define EC_MKBP_FP_MATCH_IDX_OFFSET 12 > +#define EC_MKBP_FP_MATCH_IDX_MASK 0x0000F000 > +#define EC_MKBP_FP_MATCH_IDX(fpe) (((fpe) & EC_MKBP_FP_MATCH_IDX_MASK) \ > + >> EC_MKBP_FP_MATCH_IDX_OFFSET) > +#define EC_MKBP_FP_ENROLL (1 << 27) > +#define EC_MKBP_FP_MATCH (1 << 28) > +#define EC_MKBP_FP_FINGER_DOWN (1 << 29) > +#define EC_MKBP_FP_FINGER_UP (1 << 30) > +#define EC_MKBP_FP_IMAGE_READY (1 << 31) > +/* code given by EC_MKBP_FP_ERRCODE() when EC_MKBP_FP_ENROLL is set */ > +#define EC_MKBP_FP_ERR_ENROLL_OK 0 > +#define EC_MKBP_FP_ERR_ENROLL_LOW_QUALITY 1 > +#define EC_MKBP_FP_ERR_ENROLL_IMMOBILE 2 > +#define EC_MKBP_FP_ERR_ENROLL_LOW_COVERAGE 3 > +#define EC_MKBP_FP_ERR_ENROLL_INTERNAL 5 > +/* Can be used to detect if image was usable for enrollment or not. */ > +#define EC_MKBP_FP_ERR_ENROLL_PROBLEM_MASK 1 > +/* code given by EC_MKBP_FP_ERRCODE() when EC_MKBP_FP_MATCH is set */ > +#define EC_MKBP_FP_ERR_MATCH_NO 0 > +#define EC_MKBP_FP_ERR_MATCH_NO_INTERNAL 6 > +#define EC_MKBP_FP_ERR_MATCH_NO_TEMPLATES 7 > +#define EC_MKBP_FP_ERR_MATCH_NO_LOW_QUALITY 2 > +#define EC_MKBP_FP_ERR_MATCH_NO_LOW_COVERAGE 4 > +#define EC_MKBP_FP_ERR_MATCH_YES 1 > +#define EC_MKBP_FP_ERR_MATCH_YES_UPDATED 3 > +#define EC_MKBP_FP_ERR_MATCH_YES_UPDATE_FAILED 5 > + > + > /*****************************************************************************/ > /* Temperature sensor commands */ > > /* Read temperature sensor info */ > -#define EC_CMD_TEMP_SENSOR_GET_INFO 0x70 > +#define EC_CMD_TEMP_SENSOR_GET_INFO 0x0070 > > struct ec_params_temp_sensor_get_info { > uint8_t id; > -} __packed; > +} __ec_align1; > > struct ec_response_temp_sensor_get_info { > char sensor_name[32]; > uint8_t sensor_type; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > > @@ -2245,49 +3583,131 @@ struct ec_response_temp_sensor_get_info { > /*****************************************************************************/ > /* Host event commands */ > > + > +/* Obsolete. New implementation should use EC_CMD_HOST_EVENT instead */ > /* > * Host event mask params and response structures, shared by all of the host > * event commands below. > */ > struct ec_params_host_event_mask { > uint32_t mask; > -} __packed; > +} __ec_align4; > > struct ec_response_host_event_mask { > uint32_t mask; > -} __packed; > +} __ec_align4; > > /* These all use ec_response_host_event_mask */ > -#define EC_CMD_HOST_EVENT_GET_B 0x87 > -#define EC_CMD_HOST_EVENT_GET_SMI_MASK 0x88 > -#define EC_CMD_HOST_EVENT_GET_SCI_MASK 0x89 > -#define EC_CMD_HOST_EVENT_GET_WAKE_MASK 0x8d > +#define EC_CMD_HOST_EVENT_GET_B 0x0087 > +#define EC_CMD_HOST_EVENT_GET_SMI_MASK 0x0088 > +#define EC_CMD_HOST_EVENT_GET_SCI_MASK 0x0089 > +#define EC_CMD_HOST_EVENT_GET_WAKE_MASK 0x008D > > /* These all use ec_params_host_event_mask */ > -#define EC_CMD_HOST_EVENT_SET_SMI_MASK 0x8a > -#define EC_CMD_HOST_EVENT_SET_SCI_MASK 0x8b > -#define EC_CMD_HOST_EVENT_CLEAR 0x8c > -#define EC_CMD_HOST_EVENT_SET_WAKE_MASK 0x8e > -#define EC_CMD_HOST_EVENT_CLEAR_B 0x8f > +#define EC_CMD_HOST_EVENT_SET_SMI_MASK 0x008A > +#define EC_CMD_HOST_EVENT_SET_SCI_MASK 0x008B > +#define EC_CMD_HOST_EVENT_CLEAR 0x008C > +#define EC_CMD_HOST_EVENT_SET_WAKE_MASK 0x008E > +#define EC_CMD_HOST_EVENT_CLEAR_B 0x008F > + > +/* > + * Unified host event programming interface - Should be used by newer versions > + * of BIOS/OS to program host events and masks > + */ > + > +struct ec_params_host_event { > + > + /* Action requested by host - one of enum ec_host_event_action. */ > + uint8_t action; > + > + /* > + * Mask type that the host requested the action on - one of > + * enum ec_host_event_mask_type. > + */ > + uint8_t mask_type; > + > + /* Set to 0, ignore on read */ > + uint16_t reserved; > + > + /* Value to be used in case of set operations. */ > + uint64_t value; > +} __ec_align4; > + > +/* > + * Response structure returned by EC_CMD_HOST_EVENT. > + * Update the value on a GET request. Set to 0 on GET/CLEAR > + */ > + > +struct ec_response_host_event { > + > + /* Mask value in case of get operation */ > + uint64_t value; > +} __ec_align4; > + > +enum ec_host_event_action { > + /* > + * params.value is ignored. Value of mask_type populated > + * in response.value > + */ > + EC_HOST_EVENT_GET, > + > + /* Bits in params.value are set */ > + EC_HOST_EVENT_SET, > + > + /* Bits in params.value are cleared */ > + EC_HOST_EVENT_CLEAR, > +}; > + > +enum ec_host_event_mask_type { > + > + /* Main host event copy */ > + EC_HOST_EVENT_MAIN, > + > + /* Copy B of host events */ > + EC_HOST_EVENT_B, > + > + /* SCI Mask */ > + EC_HOST_EVENT_SCI_MASK, > + > + /* SMI Mask */ > + EC_HOST_EVENT_SMI_MASK, > + > + /* Mask of events that should be always reported in hostevents */ > + EC_HOST_EVENT_ALWAYS_REPORT_MASK, > + > + /* Active wake mask */ > + EC_HOST_EVENT_ACTIVE_WAKE_MASK, > + > + /* Lazy wake mask for S0ix */ > + EC_HOST_EVENT_LAZY_WAKE_MASK_S0IX, > + > + /* Lazy wake mask for S3 */ > + EC_HOST_EVENT_LAZY_WAKE_MASK_S3, > + > + /* Lazy wake mask for S5 */ > + EC_HOST_EVENT_LAZY_WAKE_MASK_S5, > +}; > + > +#define EC_CMD_HOST_EVENT 0x00A4 > > /*****************************************************************************/ > /* Switch commands */ > > /* Enable/disable LCD backlight */ > -#define EC_CMD_SWITCH_ENABLE_BKLIGHT 0x90 > +#define EC_CMD_SWITCH_ENABLE_BKLIGHT 0x0090 > > struct ec_params_switch_enable_backlight { > uint8_t enabled; > -} __packed; > +} __ec_align1; > > /* Enable/disable WLAN/Bluetooth */ > -#define EC_CMD_SWITCH_ENABLE_WIRELESS 0x91 > +#define EC_CMD_SWITCH_ENABLE_WIRELESS 0x0091 > #define EC_VER_SWITCH_ENABLE_WIRELESS 1 > > /* Version 0 params; no response */ > struct ec_params_switch_enable_wireless_v0 { > uint8_t enabled; > -} __packed; > +} __ec_align1; > > /* Version 1 params */ > struct ec_params_switch_enable_wireless_v1 { > @@ -2306,7 +3726,7 @@ struct ec_params_switch_enable_wireless_v1 { > > /* Which flags to copy from suspend_flags */ > uint8_t suspend_mask; > -} __packed; > +} __ec_align1; > > /* Version 1 response */ > struct ec_response_switch_enable_wireless_v1 { > @@ -2315,55 +3735,56 @@ struct ec_response_switch_enable_wireless_v1 { > > /* Flags to leave enabled in S3 */ > uint8_t suspend_flags; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* GPIO commands. Only available on EC if write protect has been disabled. */ > > /* Set GPIO output value */ > -#define EC_CMD_GPIO_SET 0x92 > +#define EC_CMD_GPIO_SET 0x0092 > > struct ec_params_gpio_set { > char name[32]; > uint8_t val; > -} __packed; > +} __ec_align1; > > /* Get GPIO value */ > -#define EC_CMD_GPIO_GET 0x93 > +#define EC_CMD_GPIO_GET 0x0093 > > /* Version 0 of input params and response */ > struct ec_params_gpio_get { > char name[32]; > -} __packed; > +} __ec_align1; > + > struct ec_response_gpio_get { > uint8_t val; > -} __packed; > +} __ec_align1; > > /* Version 1 of input params and response */ > struct ec_params_gpio_get_v1 { > uint8_t subcmd; > union { > - struct { > + struct __ec_align1 { > char name[32]; > } get_value_by_name; > - struct { > + struct __ec_align1 { > uint8_t index; > } get_info; > }; > -} __packed; > +} __ec_align1; > > struct ec_response_gpio_get_v1 { > union { > - struct { > + struct __ec_align1 { > uint8_t val; > } get_value_by_name, get_count; > - struct { > + struct __ec_todo_unpacked { > uint8_t val; > char name[32]; > uint32_t flags; > } get_info; > }; > -} __packed; > +} __ec_todo_packed; > > enum gpio_get_subcmd { > EC_GPIO_GET_BY_NAME = 0, > @@ -2375,25 +3796,28 @@ enum gpio_get_subcmd { > /* I2C commands. Only available when flash write protect is unlocked. */ > > /* > - * TODO(crosbug.com/p/23570): These commands are deprecated, and will be > - * removed soon. Use EC_CMD_I2C_XFER instead. > + * CAUTION: These commands are deprecated, and are not supported anymore in EC > + * builds >= 8398.0.0 (see crosbug.com/p/23570). > + * > + * Use EC_CMD_I2C_PASSTHRU instead. > */ > > /* Read I2C bus */ > -#define EC_CMD_I2C_READ 0x94 > +#define EC_CMD_I2C_READ 0x0094 > > struct ec_params_i2c_read { > uint16_t addr; /* 8-bit address (7-bit shifted << 1) */ > uint8_t read_size; /* Either 8 or 16. */ > uint8_t port; > uint8_t offset; > -} __packed; > +} __ec_align_size1; > + > struct ec_response_i2c_read { > uint16_t data; > -} __packed; > +} __ec_align2; > > /* Write I2C bus */ > -#define EC_CMD_I2C_WRITE 0x95 > +#define EC_CMD_I2C_WRITE 0x0095 > > struct ec_params_i2c_write { > uint16_t data; > @@ -2401,7 +3825,7 @@ struct ec_params_i2c_write { > uint8_t write_size; /* Either 8 or 16. */ > uint8_t port; > uint8_t offset; > -} __packed; > +} __ec_align_size1; > > /*****************************************************************************/ > /* Charge state commands. Only available when flash write protect unlocked. */ > @@ -2409,7 +3833,7 @@ struct ec_params_i2c_write { > /* Force charge state machine to stop charging the battery or force it to > * discharge the battery. > */ > -#define EC_CMD_CHARGE_CONTROL 0x96 > +#define EC_CMD_CHARGE_CONTROL 0x0096 > #define EC_VER_CHARGE_CONTROL 1 > > enum ec_charge_control_mode { > @@ -2420,13 +3844,12 @@ enum ec_charge_control_mode { > > struct ec_params_charge_control { > uint32_t mode; /* enum charge_control_mode */ > -} __packed; > +} __ec_align4; > > /*****************************************************************************/ > -/* Console commands. Only available when flash write protect is unlocked. */ > > /* Snapshot console output buffer for use by EC_CMD_CONSOLE_READ. */ > -#define EC_CMD_CONSOLE_SNAPSHOT 0x97 > +#define EC_CMD_CONSOLE_SNAPSHOT 0x0097 > > /* > * Read data from the saved snapshot. If the subcmd parameter is > @@ -2440,7 +3863,7 @@ struct ec_params_charge_control { > * Response is null-terminated string. Empty string, if there is no more > * remaining output. > */ > -#define EC_CMD_CONSOLE_READ 0x98 > +#define EC_CMD_CONSOLE_READ 0x0098 > > enum ec_console_read_subcmd { > CONSOLE_READ_NEXT = 0, > @@ -2449,7 +3872,7 @@ enum ec_console_read_subcmd { > > struct ec_params_console_read_v1 { > uint8_t subcmd; /* enum ec_console_read_subcmd */ > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > > @@ -2460,14 +3883,13 @@ struct ec_params_console_read_v1 { > * EC_RES_SUCCESS if the command was successful. > * EC_RES_ERROR if the cut off command failed. > */ > - > -#define EC_CMD_BATTERY_CUT_OFF 0x99 > +#define EC_CMD_BATTERY_CUT_OFF 0x0099 > > #define EC_BATTERY_CUTOFF_FLAG_AT_SHUTDOWN (1 << 0) > > struct ec_params_battery_cutoff { > uint8_t flags; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* USB port mux control. */ > @@ -2475,11 +3897,11 @@ struct ec_params_battery_cutoff { > /* > * Switch USB mux or return to automatic switching. > */ > -#define EC_CMD_USB_MUX 0x9a > +#define EC_CMD_USB_MUX 0x009A > > struct ec_params_usb_mux { > uint8_t mux; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* LDOs / FETs control. */ > @@ -2492,25 +3914,25 @@ enum ec_ldo_state { > /* > * Switch on/off a LDO. > */ > -#define EC_CMD_LDO_SET 0x9b > +#define EC_CMD_LDO_SET 0x009B > > struct ec_params_ldo_set { > uint8_t index; > uint8_t state; > -} __packed; > +} __ec_align1; > > /* > * Get LDO state. > */ > -#define EC_CMD_LDO_GET 0x9c > +#define EC_CMD_LDO_GET 0x009C > > struct ec_params_ldo_get { > uint8_t index; > -} __packed; > +} __ec_align1; > > struct ec_response_ldo_get { > uint8_t state; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* Power info. */ > @@ -2518,7 +3940,7 @@ struct ec_response_ldo_get { > /* > * Get power info. > */ > -#define EC_CMD_POWER_INFO 0x9d > +#define EC_CMD_POWER_INFO 0x009D > > struct ec_response_power_info { > uint32_t usb_dev_type; > @@ -2526,12 +3948,12 @@ struct ec_response_power_info { > uint16_t voltage_system; > uint16_t current_system; > uint16_t usb_current_limit; > -} __packed; > +} __ec_align4; > > /*****************************************************************************/ > /* I2C passthru command */ > > -#define EC_CMD_I2C_PASSTHRU 0x9e > +#define EC_CMD_I2C_PASSTHRU 0x009E > > /* Read data; if not present, message is a write */ > #define EC_I2C_FLAG_READ (1 << 15) > @@ -2548,25 +3970,25 @@ struct ec_response_power_info { > struct ec_params_i2c_passthru_msg { > uint16_t addr_flags; /* I2C slave address (7 or 10 bits) and flags */ > uint16_t len; /* Number of bytes to read or write */ > -} __packed; > +} __ec_align2; > > struct ec_params_i2c_passthru { > uint8_t port; /* I2C port number */ > uint8_t num_msgs; /* Number of messages */ > struct ec_params_i2c_passthru_msg msg[]; > /* Data to write for all messages is concatenated here */ > -} __packed; > +} __ec_align2; > > struct ec_response_i2c_passthru { > uint8_t i2c_status; /* Status flags (EC_I2C_STATUS_...) */ > uint8_t num_msgs; /* Number of messages processed */ > uint8_t data[]; /* Data read by messages concatenated here */ > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* Power button hang detect */ > > -#define EC_CMD_HANG_DETECT 0x9f > +#define EC_CMD_HANG_DETECT 0x009F > > /* Reasons to start hang detection timer */ > /* Power button pressed */ > @@ -2616,7 +4038,7 @@ struct ec_params_hang_detect { > > /* Timeout in msec before generating warm reboot, if enabled */ > uint16_t warm_reboot_timeout_msec; > -} __packed; > +} __ec_align4; > > /*****************************************************************************/ > /* Commands for battery charging */ > @@ -2625,7 +4047,7 @@ struct ec_params_hang_detect { > * This is the single catch-all host command to exchange data regarding the > * charge state machine (v2 and up). > */ > -#define EC_CMD_CHARGE_STATE 0xa0 > +#define EC_CMD_CHARGE_STATE 0x00A0 > > /* Subcommands for this host command */ > enum charge_state_command { > @@ -2645,6 +4067,11 @@ enum charge_state_params { > CS_PARAM_CHG_INPUT_CURRENT, /* charger input current limit */ > CS_PARAM_CHG_STATUS, /* charger-specific status */ > CS_PARAM_CHG_OPTION, /* charger-specific options */ > + CS_PARAM_LIMIT_POWER, /* > + * Check if power is limited due to > + * low battery and / or a weak external > + * charger. READ ONLY. > + */ > /* How many so far? */ > CS_NUM_BASE_PARAMS, > > @@ -2652,30 +4079,39 @@ enum charge_state_params { > CS_PARAM_CUSTOM_PROFILE_MIN = 0x10000, > CS_PARAM_CUSTOM_PROFILE_MAX = 0x1ffff, > > + /* Range for CONFIG_CHARGE_STATE_DEBUG params */ > + CS_PARAM_DEBUG_MIN = 0x20000, > + CS_PARAM_DEBUG_CTL_MODE = 0x20000, > + CS_PARAM_DEBUG_MANUAL_MODE, > + CS_PARAM_DEBUG_SEEMS_DEAD, > + CS_PARAM_DEBUG_SEEMS_DISCONNECTED, > + CS_PARAM_DEBUG_BATT_REMOVED, > + CS_PARAM_DEBUG_MANUAL_CURRENT, > + CS_PARAM_DEBUG_MANUAL_VOLTAGE, > + CS_PARAM_DEBUG_MAX = 0x2ffff, > + > /* Other custom param ranges go here... */ > }; > > struct ec_params_charge_state { > uint8_t cmd; /* enum charge_state_command */ > union { > - struct { > - /* no args */ > - } get_state; > + /* get_state has no args */ > > - struct { > + struct __ec_todo_unpacked { > uint32_t param; /* enum charge_state_param */ > } get_param; > > - struct { > + struct __ec_todo_unpacked { > uint32_t param; /* param to set */ > uint32_t value; /* value to set */ > } set_param; > }; > -} __packed; > +} __ec_todo_packed; > > struct ec_response_charge_state { > union { > - struct { > + struct __ec_align4 { > int ac; > int chg_voltage; > int chg_current; > @@ -2683,24 +4119,23 @@ struct ec_response_charge_state { > int batt_state_of_charge; > } get_state; > > - struct { > + struct __ec_align4 { > uint32_t value; > } get_param; > - struct { > - /* no return values */ > - } set_param; > + > + /* set_param returns no args */ > }; > -} __packed; > +} __ec_align4; > > > /* > * Set maximum battery charging current. > */ > -#define EC_CMD_CHARGE_CURRENT_LIMIT 0xa1 > +#define EC_CMD_CHARGE_CURRENT_LIMIT 0x00A1 > > struct ec_params_current_limit { > uint32_t limit; /* in mA */ > -} __packed; > +} __ec_align4; > > /* > * Set maximum external voltage / current. > @@ -2711,58 +4146,134 @@ struct ec_params_current_limit { > struct ec_params_external_power_limit_v1 { > uint16_t current_lim; /* in mA, or EC_POWER_LIMIT_NONE to clear limit */ > uint16_t voltage_lim; /* in mV, or EC_POWER_LIMIT_NONE to clear limit */ > -} __packed; > +} __ec_align2; > > #define EC_POWER_LIMIT_NONE 0xffff > > +/* > + * Set maximum voltage & current of a dedicated charge port > + */ > +#define EC_CMD_OVERRIDE_DEDICATED_CHARGER_LIMIT 0x00A3 > + > +struct ec_params_dedicated_charger_limit { > + uint16_t current_lim; /* in mA */ > + uint16_t voltage_lim; /* in mV */ > +} __ec_align2; > + > +/*****************************************************************************/ > +/* Hibernate/Deep Sleep Commands */ > + > +/* Set the delay before going into hibernation. */ > +#define EC_CMD_HIBERNATION_DELAY 0x00A8 > + > +struct ec_params_hibernation_delay { > + /* > + * Seconds to wait in G3 before hibernate. Pass in 0 to read the > + * current settings without changing them. > + */ > + uint32_t seconds; > +} __ec_align4; > + > +struct ec_response_hibernation_delay { > + /* > + * The current time in seconds in which the system has been in the G3 > + * state. This value is reset if the EC transitions out of G3. > + */ > + uint32_t time_g3; > + > + /* > + * The current time remaining in seconds until the EC should hibernate. > + * This value is also reset if the EC transitions out of G3. > + */ > + uint32_t time_remaining; > + > + /* > + * The current time in seconds that the EC should wait in G3 before > + * hibernating. > + */ > + uint32_t hibernate_delay; > +} __ec_align4; > + > /* Inform the EC when entering a sleep state */ > -#define EC_CMD_HOST_SLEEP_EVENT 0xa9 > +#define EC_CMD_HOST_SLEEP_EVENT 0x00A9 > > enum host_sleep_event { > HOST_SLEEP_EVENT_S3_SUSPEND = 1, > HOST_SLEEP_EVENT_S3_RESUME = 2, > HOST_SLEEP_EVENT_S0IX_SUSPEND = 3, > - HOST_SLEEP_EVENT_S0IX_RESUME = 4 > + HOST_SLEEP_EVENT_S0IX_RESUME = 4, > + /* S3 suspend with additional enabled wake sources */ > + HOST_SLEEP_EVENT_S3_WAKEABLE_SUSPEND = 5, > }; > > struct ec_params_host_sleep_event { > uint8_t sleep_event; > -} __packed; > +} __ec_align1; > + > +/*****************************************************************************/ > +/* Device events */ > +#define EC_CMD_DEVICE_EVENT 0x00AA > + > +enum ec_device_event { > + EC_DEVICE_EVENT_TRACKPAD, > + EC_DEVICE_EVENT_DSP, > + EC_DEVICE_EVENT_WIFI, > +}; > + > +enum ec_device_event_param { > + /* Get and clear pending device events */ > + EC_DEVICE_EVENT_PARAM_GET_CURRENT_EVENTS, > + /* Get device event mask */ > + EC_DEVICE_EVENT_PARAM_GET_ENABLED_EVENTS, > + /* Set device event mask */ > + EC_DEVICE_EVENT_PARAM_SET_ENABLED_EVENTS, > +}; > + > +#define EC_DEVICE_EVENT_MASK(event_code) (1UL << (event_code % 32)) > + > +struct ec_params_device_event { > + uint32_t event_mask; > + uint8_t param; > +} __ec_align_size1; > + > +struct ec_response_device_event { > + uint32_t event_mask; > +} __ec_align4; > > /*****************************************************************************/ > /* Smart battery pass-through */ > > /* Get / Set 16-bit smart battery registers */ > -#define EC_CMD_SB_READ_WORD 0xb0 > -#define EC_CMD_SB_WRITE_WORD 0xb1 > +#define EC_CMD_SB_READ_WORD 0x00B0 > +#define EC_CMD_SB_WRITE_WORD 0x00B1 > > /* Get / Set string smart battery parameters > * formatted as SMBUS "block". > */ > -#define EC_CMD_SB_READ_BLOCK 0xb2 > -#define EC_CMD_SB_WRITE_BLOCK 0xb3 > +#define EC_CMD_SB_READ_BLOCK 0x00B2 > +#define EC_CMD_SB_WRITE_BLOCK 0x00B3 > > struct ec_params_sb_rd { > uint8_t reg; > -} __packed; > +} __ec_align1; > > struct ec_response_sb_rd_word { > uint16_t value; > -} __packed; > +} __ec_align2; > > struct ec_params_sb_wr_word { > uint8_t reg; > uint16_t value; > -} __packed; > +} __ec_align1; > > struct ec_response_sb_rd_block { > uint8_t data[32]; > -} __packed; > +} __ec_align1; > > struct ec_params_sb_wr_block { > uint8_t reg; > uint16_t data[32]; > -} __packed; > +} __ec_align1; > > /*****************************************************************************/ > /* Battery vendor parameters > @@ -2773,7 +4284,7 @@ struct ec_params_sb_wr_block { > * requested value. > */ > > -#define EC_CMD_BATTERY_VENDOR_PARAM 0xb4 > +#define EC_CMD_BATTERY_VENDOR_PARAM 0x00B4 > > enum ec_battery_vendor_param_mode { > BATTERY_VENDOR_PARAM_MODE_GET = 0, > @@ -2784,257 +4295,105 @@ struct ec_params_battery_vendor_param { > uint32_t param; > uint32_t value; > uint8_t mode; > -} __packed; > +} __ec_align_size1; > > struct ec_response_battery_vendor_param { > uint32_t value; > -} __packed; > +} __ec_align4; > > /*****************************************************************************/ > -/* Commands for I2S recording on audio codec. */ > - > -#define EC_CMD_CODEC_I2S 0x00BC > - > -enum ec_codec_i2s_subcmd { > - EC_CODEC_SET_SAMPLE_DEPTH = 0x0, > - EC_CODEC_SET_GAIN = 0x1, > - EC_CODEC_GET_GAIN = 0x2, > - EC_CODEC_I2S_ENABLE = 0x3, > - EC_CODEC_I2S_SET_CONFIG = 0x4, > - EC_CODEC_I2S_SET_TDM_CONFIG = 0x5, > - EC_CODEC_I2S_SET_BCLK = 0x6, > +/* > + * Smart Battery Firmware Update Commands > + */ > +#define EC_CMD_SB_FW_UPDATE 0x00B5 > + > +enum ec_sb_fw_update_subcmd { > + EC_SB_FW_UPDATE_PREPARE = 0x0, > + EC_SB_FW_UPDATE_INFO = 0x1, /*query sb info */ > + EC_SB_FW_UPDATE_BEGIN = 0x2, /*check if protected */ > + EC_SB_FW_UPDATE_WRITE = 0x3, /*check if protected */ > + EC_SB_FW_UPDATE_END = 0x4, > + EC_SB_FW_UPDATE_STATUS = 0x5, > + EC_SB_FW_UPDATE_PROTECT = 0x6, > + EC_SB_FW_UPDATE_MAX = 0x7, > }; > > -enum ec_sample_depth_value { > - EC_CODEC_SAMPLE_DEPTH_16 = 0, > - EC_CODEC_SAMPLE_DEPTH_24 = 1, > -}; > +#define SB_FW_UPDATE_CMD_WRITE_BLOCK_SIZE 32 > +#define SB_FW_UPDATE_CMD_STATUS_SIZE 2 > +#define SB_FW_UPDATE_CMD_INFO_SIZE 8 > > -enum ec_i2s_config { > - EC_DAI_FMT_I2S = 0, > - EC_DAI_FMT_RIGHT_J = 1, > - EC_DAI_FMT_LEFT_J = 2, > - EC_DAI_FMT_PCM_A = 3, > - EC_DAI_FMT_PCM_B = 4, > - EC_DAI_FMT_PCM_TDM = 5, > -}; > +struct ec_sb_fw_update_header { > + uint16_t subcmd; /* enum ec_sb_fw_update_subcmd */ > + uint16_t fw_id; /* firmware id */ > +} __ec_align4; > > -struct ec_param_codec_i2s { > - /* > - * enum ec_codec_i2s_subcmd > - */ > - uint8_t cmd; > +struct ec_params_sb_fw_update { > + struct ec_sb_fw_update_header hdr; > union { > - /* > - * EC_CODEC_SET_SAMPLE_DEPTH > - * Value should be one of ec_sample_depth_value. > - */ > - uint8_t depth; > - > - /* > - * EC_CODEC_SET_GAIN > - * Value should be 0~43 for both channels. > - */ > - struct ec_param_codec_i2s_set_gain { > - uint8_t left; > - uint8_t right; > - } __packed gain; > - > - /* > - * EC_CODEC_I2S_ENABLE > - * 1 to enable, 0 to disable. > - */ > - uint8_t i2s_enable; > - > - /* > - * EC_CODEC_I2S_SET_COFNIG > - * Value should be one of ec_i2s_config. > - */ > - uint8_t i2s_config; > + /* EC_SB_FW_UPDATE_PREPARE = 0x0 */ > + /* EC_SB_FW_UPDATE_INFO = 0x1 */ > + /* EC_SB_FW_UPDATE_BEGIN = 0x2 */ > + /* EC_SB_FW_UPDATE_END = 0x4 */ > + /* EC_SB_FW_UPDATE_STATUS = 0x5 */ > + /* EC_SB_FW_UPDATE_PROTECT = 0x6 */ > + /* Those have no args */ > + > + /* EC_SB_FW_UPDATE_WRITE = 0x3 */ > + struct __ec_align4 { > + uint8_t data[SB_FW_UPDATE_CMD_WRITE_BLOCK_SIZE]; > + } write; > + }; > +} __ec_align4; > > - /* > - * EC_CODEC_I2S_SET_TDM_CONFIG > - * Value should be one of ec_i2s_config. > - */ > - struct ec_param_codec_i2s_tdm { > - /* > - * 0 to 496 > - */ > - int16_t ch0_delay; > - /* > - * -1 to 496 > - */ > - int16_t ch1_delay; > - uint8_t adjacent_to_ch0; > - uint8_t adjacent_to_ch1; > - } __packed tdm_param; > +struct ec_response_sb_fw_update { > + union { > + /* EC_SB_FW_UPDATE_INFO = 0x1 */ > + struct __ec_align1 { > + uint8_t data[SB_FW_UPDATE_CMD_INFO_SIZE]; > + } info; > > - /* > - * EC_CODEC_I2S_SET_BCLK > - */ > - uint32_t bclk; > + /* EC_SB_FW_UPDATE_STATUS = 0x5 */ > + struct __ec_align1 { > + uint8_t data[SB_FW_UPDATE_CMD_STATUS_SIZE]; > + } status; > }; > -} __packed; > +} __ec_align1; > > /* > - * For subcommand EC_CODEC_GET_GAIN. > + * Entering Verified Boot Mode Command > + * Default mode is VBOOT_MODE_NORMAL if EC did not receive this command. > + * Valid Modes are: normal, developer, and recovery. > */ > -struct ec_codec_i2s_gain { > - uint8_t left; > - uint8_t right; > -} __packed; > +#define EC_CMD_ENTERING_MODE 0x00B6 > > -/*****************************************************************************/ > -/* System commands */ > +struct ec_params_entering_mode { > + int vboot_mode; > +} __ec_align4; > + > +#define VBOOT_MODE_NORMAL 0 > +#define VBOOT_MODE_DEVELOPER 1 > +#define VBOOT_MODE_RECOVERY 2 > > +/*****************************************************************************/ > /* > - * TODO(crosbug.com/p/23747): This is a confusing name, since it doesn't > - * necessarily reboot the EC. Rename to "image" or something similar? > + * I2C passthru protection command: Protects I2C tunnels against access on > + * certain addresses (board-specific). > */ > -#define EC_CMD_REBOOT_EC 0xd2 > +#define EC_CMD_I2C_PASSTHRU_PROTECT 0x00B7 > > -/* Command */ > -enum ec_reboot_cmd { > - EC_REBOOT_CANCEL = 0, /* Cancel a pending reboot */ > - EC_REBOOT_JUMP_RO = 1, /* Jump to RO without rebooting */ > - EC_REBOOT_JUMP_RW = 2, /* Jump to RW without rebooting */ > - /* (command 3 was jump to RW-B) */ > - EC_REBOOT_COLD = 4, /* Cold-reboot */ > - EC_REBOOT_DISABLE_JUMP = 5, /* Disable jump until next reboot */ > - EC_REBOOT_HIBERNATE = 6 /* Hibernate EC */ > +enum ec_i2c_passthru_protect_subcmd { > + EC_CMD_I2C_PASSTHRU_PROTECT_STATUS = 0x0, > + EC_CMD_I2C_PASSTHRU_PROTECT_ENABLE = 0x1, > }; > > -/* Flags for ec_params_reboot_ec.reboot_flags */ > -#define EC_REBOOT_FLAG_RESERVED0 (1 << 0) /* Was recovery request */ > -#define EC_REBOOT_FLAG_ON_AP_SHUTDOWN (1 << 1) /* Reboot after AP shutdown */ > - > -struct ec_params_reboot_ec { > - uint8_t cmd; /* enum ec_reboot_cmd */ > - uint8_t flags; /* See EC_REBOOT_FLAG_* */ > -} __packed; > - > -/* > - * Get information on last EC panic. > - * > - * Returns variable-length platform-dependent panic information. See panic.h > - * for details. > - */ > -#define EC_CMD_GET_PANIC_INFO 0xd3 > - > -/*****************************************************************************/ > -/* > - * ACPI commands > - * > - * These are valid ONLY on the ACPI command/data port. > - */ > - > -/* > - * ACPI Read Embedded Controller > - * > - * This reads from ACPI memory space on the EC (EC_ACPI_MEM_*). > - * > - * Use the following sequence: > - * > - * - Write EC_CMD_ACPI_READ to EC_LPC_ADDR_ACPI_CMD > - * - Wait for EC_LPC_CMDR_PENDING bit to clear > - * - Write address to EC_LPC_ADDR_ACPI_DATA > - * - Wait for EC_LPC_CMDR_DATA bit to set > - * - Read value from EC_LPC_ADDR_ACPI_DATA > - */ > -#define EC_CMD_ACPI_READ 0x80 > - > -/* > - * ACPI Write Embedded Controller > - * > - * This reads from ACPI memory space on the EC (EC_ACPI_MEM_*). > - * > - * Use the following sequence: > - * > - * - Write EC_CMD_ACPI_WRITE to EC_LPC_ADDR_ACPI_CMD > - * - Wait for EC_LPC_CMDR_PENDING bit to clear > - * - Write address to EC_LPC_ADDR_ACPI_DATA > - * - Wait for EC_LPC_CMDR_PENDING bit to clear > - * - Write value to EC_LPC_ADDR_ACPI_DATA > - */ > -#define EC_CMD_ACPI_WRITE 0x81 > - > -/* > - * ACPI Query Embedded Controller > - * > - * This clears the lowest-order bit in the currently pending host events, and > - * sets the result code to the 1-based index of the bit (event 0x00000001 = 1, > - * event 0x80000000 = 32), or 0 if no event was pending. > - */ > -#define EC_CMD_ACPI_QUERY_EVENT 0x84 > - > -/* Valid addresses in ACPI memory space, for read/write commands */ > - > -/* Memory space version; set to EC_ACPI_MEM_VERSION_CURRENT */ > -#define EC_ACPI_MEM_VERSION 0x00 > -/* > - * Test location; writing value here updates test compliment byte to (0xff - > - * value). > - */ > -#define EC_ACPI_MEM_TEST 0x01 > -/* Test compliment; writes here are ignored. */ > -#define EC_ACPI_MEM_TEST_COMPLIMENT 0x02 > - > -/* Keyboard backlight brightness percent (0 - 100) */ > -#define EC_ACPI_MEM_KEYBOARD_BACKLIGHT 0x03 > -/* DPTF Target Fan Duty (0-100, 0xff for auto/none) */ > -#define EC_ACPI_MEM_FAN_DUTY 0x04 > - > -/* > - * DPTF temp thresholds. Any of the EC's temp sensors can have up to two > - * independent thresholds attached to them. The current value of the ID > - * register determines which sensor is affected by the THRESHOLD and COMMIT > - * registers. The THRESHOLD register uses the same EC_TEMP_SENSOR_OFFSET scheme > - * as the memory-mapped sensors. The COMMIT register applies those settings. > - * > - * The spec does not mandate any way to read back the threshold settings > - * themselves, but when a threshold is crossed the AP needs a way to determine > - * which sensor(s) are responsible. Each reading of the ID register clears and > - * returns one sensor ID that has crossed one of its threshold (in either > - * direction) since the last read. A value of 0xFF means "no new thresholds > - * have tripped". Setting or enabling the thresholds for a sensor will clear > - * the unread event count for that sensor. > - */ > -#define EC_ACPI_MEM_TEMP_ID 0x05 > -#define EC_ACPI_MEM_TEMP_THRESHOLD 0x06 > -#define EC_ACPI_MEM_TEMP_COMMIT 0x07 > -/* > - * Here are the bits for the COMMIT register: > - * bit 0 selects the threshold index for the chosen sensor (0/1) > - * bit 1 enables/disables the selected threshold (0 = off, 1 = on) > - * Each write to the commit register affects one threshold. > - */ > -#define EC_ACPI_MEM_TEMP_COMMIT_SELECT_MASK (1 << 0) > -#define EC_ACPI_MEM_TEMP_COMMIT_ENABLE_MASK (1 << 1) > -/* > - * Example: > - * > - * Set the thresholds for sensor 2 to 50 C and 60 C: > - * write 2 to [0x05] -- select temp sensor 2 > - * write 0x7b to [0x06] -- C_TO_K(50) - EC_TEMP_SENSOR_OFFSET > - * write 0x2 to [0x07] -- enable threshold 0 with this value > - * write 0x85 to [0x06] -- C_TO_K(60) - EC_TEMP_SENSOR_OFFSET > - * write 0x3 to [0x07] -- enable threshold 1 with this value > - * > - * Disable the 60 C threshold, leaving the 50 C threshold unchanged: > - * write 2 to [0x05] -- select temp sensor 2 > - * write 0x1 to [0x07] -- disable threshold 1 > - */ > - > -/* DPTF battery charging current limit */ > -#define EC_ACPI_MEM_CHARGING_LIMIT 0x08 > - > -/* Charging limit is specified in 64 mA steps */ > -#define EC_ACPI_MEM_CHARGING_LIMIT_STEP_MA 64 > -/* Value to disable DPTF battery charging limit */ > -#define EC_ACPI_MEM_CHARGING_LIMIT_DISABLED 0xff > +struct ec_params_i2c_passthru_protect { > + uint8_t subcmd; > + uint8_t port; /* I2C port number */ > +} __ec_align1; > > -/* Current version of ACPI memory address space */ > -#define EC_ACPI_MEM_VERSION_CURRENT 1 > +struct ec_response_i2c_passthru_protect { > + uint8_t status; /* Status flags (0: unlocked, 1: locked) */ > +} __ec_align1; > > > /*****************************************************************************/ > @@ -3043,7 +4402,8 @@ struct ec_params_reboot_ec { > * > * These commands are for sending and receiving message via HDMI CEC > */ > -#define EC_MAX_CEC_MSG_LEN 16 > + > +#define MAX_CEC_MSG_LEN 16 > > /* CEC message from the AP to be written on the CEC bus */ > #define EC_CMD_CEC_WRITE_MSG 0x00B8 > @@ -3053,8 +4413,8 @@ struct ec_params_reboot_ec { > * @msg: message content to write to the CEC bus > */ > struct ec_params_cec_write { > - uint8_t msg[EC_MAX_CEC_MSG_LEN]; > -} __packed; > + uint8_t msg[MAX_CEC_MSG_LEN]; > +} __ec_align1; > > /* Set various CEC parameters */ > #define EC_CMD_CEC_SET 0x00BA > @@ -3063,14 +4423,14 @@ struct ec_params_cec_write { > * struct ec_params_cec_set - CEC parameters set > * @cmd: parameter type, can be CEC_CMD_ENABLE or CEC_CMD_LOGICAL_ADDRESS > * @val: in case cmd is CEC_CMD_ENABLE, this field can be 0 to disable CEC > - * or 1 to enable CEC functionality, in case cmd is CEC_CMD_LOGICAL_ADDRESS, > - * this field encodes the requested logical address between 0 and 15 > - * or 0xff to unregister > + * or 1 to enable CEC functionality, in case cmd is > + * CEC_CMD_LOGICAL_ADDRESS, this field encodes the requested logical > + * address between 0 and 15 or 0xff to unregister > */ > struct ec_params_cec_set { > uint8_t cmd; /* enum cec_command */ > uint8_t val; > -} __packed; > +} __ec_align1; > > /* Read various CEC parameters */ > #define EC_CMD_CEC_GET 0x00BB > @@ -3081,7 +4441,7 @@ struct ec_params_cec_set { > */ > struct ec_params_cec_get { > uint8_t cmd; /* enum cec_command */ > -} __packed; > +} __ec_align1; > > /** > * struct ec_response_cec_get - CEC parameters get response > @@ -3092,10 +4452,10 @@ struct ec_params_cec_get { > */ > struct ec_response_cec_get { > uint8_t val; > -} __packed; > +} __ec_align1; > > /* CEC parameters command */ > -enum ec_cec_command { > +enum cec_command { > /* CEC reading, writing and events enable */ > CEC_CMD_ENABLE, > /* CEC logical address */ > @@ -3105,11 +4465,140 @@ enum ec_cec_command { > /* Events from CEC to AP */ > enum mkbp_cec_event { > /* Outgoing message was acknowledged by a follower */ > - EC_MKBP_CEC_SEND_OK = BIT(0), > + EC_MKBP_CEC_SEND_OK = 1 << 0, > /* Outgoing message was not acknowledged */ > - EC_MKBP_CEC_SEND_FAILED = BIT(1), > + EC_MKBP_CEC_SEND_FAILED = 1 << 1, > +}; > + > +/*****************************************************************************/ > + > +/* Commands for I2S recording on audio codec. */ > + > +#define EC_CMD_CODEC_I2S 0x00BC > +#define EC_WOV_I2S_SAMPLE_RATE 48000 > + > +enum ec_codec_i2s_subcmd { > + EC_CODEC_SET_SAMPLE_DEPTH = 0x0, > + EC_CODEC_SET_GAIN = 0x1, > + EC_CODEC_GET_GAIN = 0x2, > + EC_CODEC_I2S_ENABLE = 0x3, > + EC_CODEC_I2S_SET_CONFIG = 0x4, > + EC_CODEC_I2S_SET_TDM_CONFIG = 0x5, > + EC_CODEC_I2S_SET_BCLK = 0x6, > + EC_CODEC_I2S_SUBCMD_COUNT = 0x7, > +}; > + > +enum ec_sample_depth_value { > + EC_CODEC_SAMPLE_DEPTH_16 = 0, > + EC_CODEC_SAMPLE_DEPTH_24 = 1, > +}; > + > +enum ec_i2s_config { > + EC_DAI_FMT_I2S = 0, > + EC_DAI_FMT_RIGHT_J = 1, > + EC_DAI_FMT_LEFT_J = 2, > + EC_DAI_FMT_PCM_A = 3, > + EC_DAI_FMT_PCM_B = 4, > + EC_DAI_FMT_PCM_TDM = 5, > +}; > + > +/* > + * For subcommand EC_CODEC_GET_GAIN. > + */ > +struct __ec_align1 ec_codec_i2s_gain { > + uint8_t left; > + uint8_t right; > +}; > + > +struct __ec_todo_unpacked ec_param_codec_i2s_tdm { > + int16_t ch0_delay; /* 0 to 496 */ > + int16_t ch1_delay; /* -1 to 496 */ > + uint8_t adjacent_to_ch0; > + uint8_t adjacent_to_ch1; > +}; > + > +struct __ec_todo_packed ec_param_codec_i2s { > + /* enum ec_codec_i2s_subcmd */ > + uint8_t cmd; > + union { > + /* > + * EC_CODEC_SET_SAMPLE_DEPTH > + * Value should be one of ec_sample_depth_value. > + */ > + uint8_t depth; > + > + /* > + * EC_CODEC_SET_GAIN > + * Value should be 0~43 for both channels. > + */ > + struct ec_codec_i2s_gain gain; > + > + /* > + * EC_CODEC_I2S_ENABLE > + * 1 to enable, 0 to disable. > + */ > + uint8_t i2s_enable; > + > + /* > + * EC_CODEC_I2S_SET_CONFIG > + * Value should be one of ec_i2s_config. > + */ > + uint8_t i2s_config; > + > + /* > + * EC_CODEC_I2S_SET_TDM_CONFIG > + * Value should be one of ec_i2s_config. > + */ > + struct ec_param_codec_i2s_tdm tdm_param; > + > + /* > + * EC_CODEC_I2S_SET_BCLK > + */ > + uint32_t bclk; > + }; > +}; > + > + > +/*****************************************************************************/ > + > +/* System commands */ > + > +/* > + * TODO(crosbug.com/p/23747): This is a confusing name, since it doesn't > + * necessarily reboot the EC. Rename to "image" or something similar? > + */ > +#define EC_CMD_REBOOT_EC 0x00D2 > + > +/* Command */ > +enum ec_reboot_cmd { > + EC_REBOOT_CANCEL = 0, /* Cancel a pending reboot */ > + EC_REBOOT_JUMP_RO = 1, /* Jump to RO without rebooting */ > + EC_REBOOT_JUMP_RW = 2, /* Jump to active RW without rebooting */ > + /* (command 3 was jump to RW-B) */ > + EC_REBOOT_COLD = 4, /* Cold-reboot */ > + EC_REBOOT_DISABLE_JUMP = 5, /* Disable jump until next reboot */ > + EC_REBOOT_HIBERNATE = 6, /* Hibernate EC */ > + EC_REBOOT_HIBERNATE_CLEAR_AP_OFF = 7, /* and clears AP_OFF flag */ > }; > > +/* Flags for ec_params_reboot_ec.reboot_flags */ > +#define EC_REBOOT_FLAG_RESERVED0 (1 << 0) /* Was recovery request */ > +#define EC_REBOOT_FLAG_ON_AP_SHUTDOWN (1 << 1) /* Reboot after AP shutdown */ > +#define EC_REBOOT_FLAG_SWITCH_RW_SLOT (1 << 2) /* Switch RW slot */ > + > +struct ec_params_reboot_ec { > + uint8_t cmd; /* enum ec_reboot_cmd */ > + uint8_t flags; /* See EC_REBOOT_FLAG_* */ > +} __ec_align1; > + > +/* > + * Get information on last EC panic. > + * > + * Returns variable-length platform-dependent panic information. See panic.h > + * for details. > + */ > +#define EC_CMD_GET_PANIC_INFO 0x00D3 > + > /*****************************************************************************/ > /* > * Special commands > @@ -3127,7 +4616,7 @@ enum mkbp_cec_event { > * > * Use EC_CMD_REBOOT_EC to reboot the EC more politely. > */ > -#define EC_CMD_REBOOT 0xd1 /* Think "die" */ > +#define EC_CMD_REBOOT 0x00D1 /* Think "die" */ > > /* > * Resend last response (not supported on LPC). > @@ -3136,7 +4625,7 @@ enum mkbp_cec_event { > * there was no previous command, or the previous command's response was too > * big to save. > */ > -#define EC_CMD_RESEND_RESPONSE 0xdb > +#define EC_CMD_RESEND_RESPONSE 0x00DB > > /* > * This header byte on a command indicate version 0. Any header byte less > @@ -3148,9 +4637,7 @@ enum mkbp_cec_event { > * > * The old EC interface must not use commands 0xdc or higher. > */ > -#define EC_CMD_VERSION0 0xdc > - > -#endif /* !__ACPI__ */ > +#define EC_CMD_VERSION0 0x00DC > > /*****************************************************************************/ > /* > @@ -3160,21 +4647,56 @@ enum mkbp_cec_event { > */ > > /* EC to PD MCU exchange status command */ > -#define EC_CMD_PD_EXCHANGE_STATUS 0x100 > +#define EC_CMD_PD_EXCHANGE_STATUS 0x0100 > +#define EC_VER_PD_EXCHANGE_STATUS 2 > + > +enum pd_charge_state { > + PD_CHARGE_NO_CHANGE = 0, /* Don't change charge state */ > + PD_CHARGE_NONE, /* No charging allowed */ > + PD_CHARGE_5V, /* 5V charging only */ > + PD_CHARGE_MAX /* Charge at max voltage */ > +}; > > /* Status of EC being sent to PD */ > +#define EC_STATUS_HIBERNATING (1 << 0) > + > struct ec_params_pd_status { > - int8_t batt_soc; /* battery state of charge */ > -} __packed; > + uint8_t status; /* EC status */ > + int8_t batt_soc; /* battery state of charge */ > + uint8_t charge_state; /* charging state (from enum pd_charge_state) */ > +} __ec_align1; > > /* Status of PD being sent back to EC */ > +#define PD_STATUS_HOST_EVENT (1 << 0) /* Forward host event to AP */ > +#define PD_STATUS_IN_RW (1 << 1) /* Running RW image */ > +#define PD_STATUS_JUMPED_TO_IMAGE (1 << 2) /* Current image was jumped to */ > +#define PD_STATUS_TCPC_ALERT_0 (1 << 3) /* Alert active in port 0 TCPC */ > +#define PD_STATUS_TCPC_ALERT_1 (1 << 4) /* Alert active in port 1 TCPC */ > +#define PD_STATUS_TCPC_ALERT_2 (1 << 5) /* Alert active in port 2 TCPC */ > +#define PD_STATUS_TCPC_ALERT_3 (1 << 6) /* Alert active in port 3 TCPC */ > +#define PD_STATUS_EC_INT_ACTIVE (PD_STATUS_TCPC_ALERT_0 | \ > + PD_STATUS_TCPC_ALERT_1 | \ > + PD_STATUS_HOST_EVENT) > struct ec_response_pd_status { > - int8_t status; /* PD MCU status */ > - uint32_t curr_lim_ma; /* input current limit */ > -} __packed; > + uint32_t curr_lim_ma; /* input current limit */ > + uint16_t status; /* PD MCU status */ > + int8_t active_charge_port; /* active charging port */ > +} __ec_align_size1; > + > +/* AP to PD MCU host event status command, cleared on read */ > +#define EC_CMD_PD_HOST_EVENT_STATUS 0x0104 > + > +/* PD MCU host event status bits */ > +#define PD_EVENT_UPDATE_DEVICE (1 << 0) > +#define PD_EVENT_POWER_CHANGE (1 << 1) > +#define PD_EVENT_IDENTITY_RECEIVED (1 << 2) > +#define PD_EVENT_DATA_SWAP (1 << 3) > +struct ec_response_host_event_status { > + uint32_t status; /* PD MCU host event status */ > +} __ec_align4; > > /* Set USB type-C port role and muxes */ > -#define EC_CMD_USB_PD_CONTROL 0x101 > +#define EC_CMD_USB_PD_CONTROL 0x0101 > > enum usb_pd_control_role { > USB_PD_CTRL_ROLE_NO_CHANGE = 0, > @@ -3182,6 +4704,8 @@ enum usb_pd_control_role { > USB_PD_CTRL_ROLE_TOGGLE_OFF = 2, > USB_PD_CTRL_ROLE_FORCE_SINK = 3, > USB_PD_CTRL_ROLE_FORCE_SOURCE = 4, > + USB_PD_CTRL_ROLE_FREEZE = 5, > + USB_PD_CTRL_ROLE_COUNT > }; > > enum usb_pd_control_mux { > @@ -3191,6 +4715,7 @@ enum usb_pd_control_mux { > USB_PD_CTRL_MUX_DP = 3, > USB_PD_CTRL_MUX_DOCK = 4, > USB_PD_CTRL_MUX_AUTO = 5, > + USB_PD_CTRL_MUX_COUNT > }; > > enum usb_pd_control_swap { > @@ -3206,42 +4731,49 @@ struct ec_params_usb_pd_control { > uint8_t role; > uint8_t mux; > uint8_t swap; > -} __packed; > +} __ec_align1; > > #define PD_CTRL_RESP_ENABLED_COMMS (1 << 0) /* Communication enabled */ > #define PD_CTRL_RESP_ENABLED_CONNECTED (1 << 1) /* Device connected */ > #define PD_CTRL_RESP_ENABLED_PD_CAPABLE (1 << 2) /* Partner is PD capable */ > > -#define PD_CTRL_RESP_ROLE_POWER BIT(0) /* 0=SNK/1=SRC */ > -#define PD_CTRL_RESP_ROLE_DATA BIT(1) /* 0=UFP/1=DFP */ > -#define PD_CTRL_RESP_ROLE_VCONN BIT(2) /* Vconn status */ > -#define PD_CTRL_RESP_ROLE_DR_POWER BIT(3) /* Partner is dualrole power */ > -#define PD_CTRL_RESP_ROLE_DR_DATA BIT(4) /* Partner is dualrole data */ > -#define PD_CTRL_RESP_ROLE_USB_COMM BIT(5) /* Partner USB comm capable */ > -#define PD_CTRL_RESP_ROLE_EXT_POWERED BIT(6) /* Partner externally powerd */ > +#define PD_CTRL_RESP_ROLE_POWER (1 << 0) /* 0=SNK/1=SRC */ > +#define PD_CTRL_RESP_ROLE_DATA (1 << 1) /* 0=UFP/1=DFP */ > +#define PD_CTRL_RESP_ROLE_VCONN (1 << 2) /* Vconn status */ > +#define PD_CTRL_RESP_ROLE_DR_POWER (1 << 3) /* Partner is > dualrole power */ > +#define PD_CTRL_RESP_ROLE_DR_DATA (1 << 4) /* Partner is dualrole data */ > +#define PD_CTRL_RESP_ROLE_USB_COMM (1 << 5) /* Partner USB comm capable */ > +#define PD_CTRL_RESP_ROLE_EXT_POWERED (1 << 6) /* Partner > externally powerd */ > + > +struct ec_response_usb_pd_control { > + uint8_t enabled; > + uint8_t role; > + uint8_t polarity; > + uint8_t state; > +} __ec_align1; > > struct ec_response_usb_pd_control_v1 { > uint8_t enabled; > uint8_t role; > uint8_t polarity; > char state[32]; > -} __packed; > +} __ec_align1; > > -#define EC_CMD_USB_PD_PORTS 0x102 > +#define EC_CMD_USB_PD_PORTS 0x0102 > > /* Maximum number of PD ports on a device, num_ports will be <= this */ > #define EC_USB_PD_MAX_PORTS 8 > > struct ec_response_usb_pd_ports { > uint8_t num_ports; > -} __packed; > +} __ec_align1; > > -#define EC_CMD_USB_PD_POWER_INFO 0x103 > +#define EC_CMD_USB_PD_POWER_INFO 0x0103 > > #define PD_POWER_CHARGING_PORT 0xff > struct ec_params_usb_pd_power_info { > uint8_t port; > -} __packed; > +} __ec_align1; > > enum usb_chg_type { > USB_CHG_TYPE_NONE, > @@ -3254,6 +4786,7 @@ enum usb_chg_type { > USB_CHG_TYPE_OTHER, > USB_CHG_TYPE_VBUS, > USB_CHG_TYPE_UNKNOWN, > + USB_CHG_TYPE_DEDICATED, > }; > enum usb_power_roles { > USB_PD_PORT_POWER_DISCONNECTED, > @@ -3267,7 +4800,7 @@ struct usb_chg_measures { > uint16_t voltage_now; > uint16_t current_max; > uint16_t current_lim; > -} __packed; > +} __ec_align2; > > struct ec_response_usb_pd_power_info { > uint8_t role; > @@ -3276,11 +4809,8 @@ struct ec_response_usb_pd_power_info { > uint8_t reserved1; > struct usb_chg_measures meas; > uint32_t max_power; > -} __packed; > +} __ec_align4; > > -struct ec_params_usb_pd_info_request { > - uint8_t port; > -} __packed; > > /* > * This command will return the number of USB PD charge port + the number > @@ -3290,7 +4820,47 @@ struct ec_params_usb_pd_info_request { > #define EC_CMD_CHARGE_PORT_COUNT 0x0105 > struct ec_response_charge_port_count { > uint8_t port_count; > -} __packed; > +} __ec_align1; > + > +/* Write USB-PD device FW */ > +#define EC_CMD_USB_PD_FW_UPDATE 0x0110 > + > +enum usb_pd_fw_update_cmds { > + USB_PD_FW_REBOOT, > + USB_PD_FW_FLASH_ERASE, > + USB_PD_FW_FLASH_WRITE, > + USB_PD_FW_ERASE_SIG, > +}; > + > +struct ec_params_usb_pd_fw_update { > + uint16_t dev_id; > + uint8_t cmd; > + uint8_t port; > + uint32_t size; /* Size to write in bytes */ > + /* Followed by data to write */ > +} __ec_align4; > + > +/* Write USB-PD Accessory RW_HASH table entry */ > +#define EC_CMD_USB_PD_RW_HASH_ENTRY 0x0111 > +/* RW hash is first 20 bytes of SHA-256 of RW section */ > +#define PD_RW_HASH_SIZE 20 > +struct ec_params_usb_pd_rw_hash_entry { > + uint16_t dev_id; > + uint8_t dev_rw_hash[PD_RW_HASH_SIZE]; > + uint8_t reserved; /* > + * For alignment of current_image > + * TODO(rspangler) but it's not aligned! > + * Should have been reserved[2]. > + */ > + uint32_t current_image; /* One of ec_current_image */ > +} __ec_align1; > + > +/* Read USB-PD Accessory info */ > +#define EC_CMD_USB_PD_DEV_INFO 0x0112 > + > +struct ec_params_usb_pd_info_request { > + uint8_t port; > +} __ec_align1; > > /* Read USB-PD Device discovery info */ > #define EC_CMD_USB_PD_DISCOVERY 0x0113 > @@ -3298,7 +4868,7 @@ struct ec_params_usb_pd_discovery_entry { > uint16_t vid; /* USB-IF VID */ > uint16_t pid; /* USB-IF PID */ > uint8_t ptype; /* product type (hub,periph,cable,ama) */ > -} __packed; > +} __ec_align_size1; > > /* Override default charge behavior */ > #define EC_CMD_PD_CHARGE_PORT_OVERRIDE 0x0114 > @@ -3312,9 +4882,13 @@ enum usb_pd_override_ports { > > struct ec_params_charge_port_override { > int16_t override_port; /* Override port# */ > -} __packed; > +} __ec_align2; > > -/* Read (and delete) one entry of PD event log */ > +/* > + * Read (and delete) one entry of PD event log. > + * TODO(crbug.com/751742): Make this host command more generic to accommodate > + * future non-PD logs that use the same internal EC event_log. > + */ > #define EC_CMD_PD_GET_LOG_ENTRY 0x0115 > > struct ec_response_pd_log { > @@ -3323,7 +4897,7 @@ struct ec_response_pd_log { > uint8_t size_port; /* [7:5] port number [4:0] payload size in bytes */ > uint16_t data; /* type-defined data payload */ > uint8_t payload[0]; /* optional additional data payload: 0..16 bytes */ > -} __packed; > +} __ec_align4; > > /* The timestamp is the microsecond counter shifted to get about a ms. */ > #define PD_LOG_TIMESTAMP_SHIFT 10 /* 1 LSB = 1024us */ > @@ -3363,11 +4937,11 @@ struct ec_response_pd_log { > * the data field contains the port state flags as defined below : > */ > /* Port partner is a dual role device */ > -#define CHARGE_FLAGS_DUAL_ROLE BIT(15) > +#define CHARGE_FLAGS_DUAL_ROLE (1 << 15) > /* Port is the pending override port */ > -#define CHARGE_FLAGS_DELAYED_OVERRIDE BIT(14) > +#define CHARGE_FLAGS_DELAYED_OVERRIDE (1 << 14) > /* Port is the override port */ > -#define CHARGE_FLAGS_OVERRIDE BIT(13) > +#define CHARGE_FLAGS_OVERRIDE (1 << 13) > /* Charger type */ > #define CHARGE_FLAGS_TYPE_SHIFT 3 > #define CHARGE_FLAGS_TYPE_MASK (0xf << CHARGE_FLAGS_TYPE_SHIFT) > @@ -3389,35 +4963,698 @@ struct mcdp_version { > uint8_t major; > uint8_t minor; > uint16_t build; > -} __packed; > +} __ec_align4; > > struct mcdp_info { > uint8_t family[2]; > uint8_t chipid[2]; > struct mcdp_version irom; > struct mcdp_version fw; > -} __packed; > +} __ec_align4; > > /* struct mcdp_info field decoding */ > #define MCDP_CHIPID(chipid) ((chipid[0] << 8) | chipid[1]) > #define MCDP_FAMILY(family) ((family[0] << 8) | family[1]) > > +/* Get/Set USB-PD Alternate mode info */ > +#define EC_CMD_USB_PD_GET_AMODE 0x0116 > +struct ec_params_usb_pd_get_mode_request { > + uint16_t svid_idx; /* SVID index to get */ > + uint8_t port; /* port */ > +} __ec_align_size1; > + > +struct ec_params_usb_pd_get_mode_response { > + uint16_t svid; /* SVID */ > + uint16_t opos; /* Object Position */ > + uint32_t vdo[6]; /* Mode VDOs */ > +} __ec_align4; > + > +#define EC_CMD_USB_PD_SET_AMODE 0x0117 > + > +enum pd_mode_cmd { > + PD_EXIT_MODE = 0, > + PD_ENTER_MODE = 1, > + /* Not a command. Do NOT remove. */ > + PD_MODE_CMD_COUNT, > +}; > + > +struct ec_params_usb_pd_set_mode_request { > + uint32_t cmd; /* enum pd_mode_cmd */ > + uint16_t svid; /* SVID to set */ > + uint8_t opos; /* Object Position */ > + uint8_t port; /* port */ > +} __ec_align4; > + > +/* Ask the PD MCU to record a log of a requested type */ > +#define EC_CMD_PD_WRITE_LOG_ENTRY 0x0118 > + > +struct ec_params_pd_write_log_entry { > + uint8_t type; /* event type : see PD_EVENT_xx above */ > + uint8_t port; /* port#, or 0 for events unrelated to a given port */ > +} __ec_align1; > + > + > +/* Control USB-PD chip */ > +#define EC_CMD_PD_CONTROL 0x0119 > + > +enum ec_pd_control_cmd { > + PD_SUSPEND = 0, /* Suspend the PD chip (EC: stop talking to PD) */ > + PD_RESUME, /* Resume the PD chip (EC: start talking to PD) */ > + PD_RESET, /* Force reset the PD chip */ > + PD_CONTROL_DISABLE, /* Disable further calls to this command */ > + PD_CHIP_ON, /* Power on the PD chip */ > +}; > + > +struct ec_params_pd_control { > + uint8_t chip; /* chip id */ > + uint8_t subcmd; > +} __ec_align1; > + > /* Get info about USB-C SS muxes */ > -#define EC_CMD_USB_PD_MUX_INFO 0x11a > +#define EC_CMD_USB_PD_MUX_INFO 0x011A > > struct ec_params_usb_pd_mux_info { > uint8_t port; /* USB-C port number */ > -} __packed; > +} __ec_align1; > > /* Flags representing mux state */ > -#define USB_PD_MUX_USB_ENABLED (1 << 0) > -#define USB_PD_MUX_DP_ENABLED (1 << 1) > -#define USB_PD_MUX_POLARITY_INVERTED (1 << 2) > -#define USB_PD_MUX_HPD_IRQ (1 << 3) > +#define USB_PD_MUX_USB_ENABLED (1 << 0) /* USB connected */ > +#define USB_PD_MUX_DP_ENABLED (1 << 1) /* DP connected */ > +#define USB_PD_MUX_POLARITY_INVERTED (1 << 2) /* CC line Polarity inverted */ > +#define USB_PD_MUX_HPD_IRQ (1 << 3) /* HPD IRQ is asserted */ > +#define USB_PD_MUX_HPD_LVL (1 << 4) /* HPD level is asserted */ > > struct ec_response_usb_pd_mux_info { > uint8_t flags; /* USB_PD_MUX_*-encoded USB mux state */ > -} __packed; > +} __ec_align1; > + > +#define EC_CMD_PD_CHIP_INFO 0x011B > + > +struct ec_params_pd_chip_info { > + uint8_t port; /* USB-C port number */ > + uint8_t renew; /* Force renewal */ > +} __ec_align1; > + > +struct ec_response_pd_chip_info { > + uint16_t vendor_id; > + uint16_t product_id; > + uint16_t device_id; > + union { > + uint8_t fw_version_string[8]; > + uint64_t fw_version_number; > + }; > +} __ec_align2; > + > +struct ec_response_pd_chip_info_v1 { > + uint16_t vendor_id; > + uint16_t product_id; > + uint16_t device_id; > + union { > + uint8_t fw_version_string[8]; > + uint64_t fw_version_number; > + }; > + union { > + uint8_t min_req_fw_version_string[8]; > + uint64_t min_req_fw_version_number; > + }; > +} __ec_align2; > + > +/* Run RW signature verification and get status */ > +#define EC_CMD_RWSIG_CHECK_STATUS 0x011C > + > +struct ec_response_rwsig_check_status { > + uint32_t status; > +} __ec_align4; > + > +/* For controlling RWSIG task */ > +#define EC_CMD_RWSIG_ACTION 0x011D > + > +enum rwsig_action { > + RWSIG_ACTION_ABORT = 0, /* Abort RWSIG and prevent jumping */ > + RWSIG_ACTION_CONTINUE = 1, /* Jump to RW immediately */ > +}; > + > +struct ec_params_rwsig_action { > + uint32_t action; > +} __ec_align4; > + > +/* Run verification on a slot */ > +#define EC_CMD_EFS_VERIFY 0x011E > + > +struct ec_params_efs_verify { > + uint8_t region; /* enum ec_flash_region */ > +} __ec_align1; > + > +/* > + * Retrieve info from Cros Board Info store. Response is based on the data > + * type. Integers return a uint32. Strings return a string, using the response > + * size to determine how big it is. > + */ > +#define EC_CMD_GET_CROS_BOARD_INFO 0x011F > +/* > + * Write info into Cros Board Info on EEPROM. Write fails if the board has > + * hardware write-protect enabled. > + */ > +#define EC_CMD_SET_CROS_BOARD_INFO 0x0120 > + > +enum cbi_data_tag { > + CBI_TAG_BOARD_VERSION = 0, /* uint32_t or smaller */ > + CBI_TAG_OEM_ID = 1, /* uint32_t or smaller */ > + CBI_TAG_SKU_ID = 2, /* uint32_t or smaller */ > + CBI_TAG_DRAM_PART_NUM = 3, /* variable length ascii, nul terminated. */ > + CBI_TAG_OEM_NAME = 4, /* variable length ascii, nul terminated. */ > + CBI_TAG_MODEL_ID = 5, /* uint32_t or smaller */ > + CBI_TAG_COUNT, > +}; > + > +/* > + * Flags to control read operation > + * > + * RELOAD: Invalidate cache and read data from EEPROM. Useful to verify > + * write was successful without reboot. > + */ > +#define CBI_GET_RELOAD (1 << 0) > + > +struct ec_params_get_cbi { > + uint32_t tag; /* enum cbi_data_tag */ > + uint32_t flag; /* CBI_GET_* */ > +} __ec_align4; > + > +/* > + * Flags to control write behavior. > + * > + * NO_SYNC: Makes EC update data in RAM but skip writing to EEPROM. It's > + * useful when writing multiple fields in a row. > + * INIT: Need to be set when creating a new CBI from scratch. All fields > + * will be initialized to zero first. > + */ > +#define CBI_SET_NO_SYNC (1 << 0) > +#define CBI_SET_INIT (1 << 1) > + > +struct ec_params_set_cbi { > + uint32_t tag; /* enum cbi_data_tag */ > + uint32_t flag; /* CBI_SET_* */ > + uint32_t size; /* Data size */ > + uint8_t data[]; /* For string and raw data */ > +} __ec_align1; > + > +/* > + * Information about resets of the AP by the EC and the EC's own uptime. > + */ > +#define EC_CMD_GET_UPTIME_INFO 0x0121 > + > +struct ec_response_uptime_info { > + /* > + * Number of milliseconds since the last EC boot. Sysjump resets > + * typically do not restart the EC's time_since_boot epoch. > + * > + * WARNING: The EC's sense of time is much less accurate than the AP's > + * sense of time, in both phase and frequency. This timebase is similar > + * to CLOCK_MONOTONIC_RAW, but with 1% or more frequency error. > + */ > + uint32_t time_since_ec_boot_ms; > + > + /* > + * Number of times the AP was reset by the EC since the last EC boot. > + * Note that the AP may be held in reset by the EC during the initial > + * boot sequence, such that the very first AP boot may count as more > + * than one here. > + */ > + uint32_t ap_resets_since_ec_boot; > + > + /* > + * The set of flags which describe the EC's most recent reset. See > + * include/system.h RESET_FLAG_* for details. > + */ > + uint32_t ec_reset_flags; > + > + /* Empty log entries have both the cause and timestamp set to zero. */ > + struct ap_reset_log_entry { > + /* > + * See include/chipset.h: enum chipset_{reset,shutdown}_reason > + * for details. > + */ > + uint16_t reset_cause; > + > + /* Reserved for protocol growth. */ > + uint16_t reserved; > + > + /* > + * The time of the reset's assertion, in milliseconds since the > + * last EC boot, in the same epoch as time_since_ec_boot_ms. > + * Set to zero if the log entry is empty. > + */ > + uint32_t reset_time_ms; > + } recent_ap_reset[4]; > +} __ec_align4; > + > +/* > + * Add entropy to the device secret (stored in the rollback region). > + * > + * Depending on the chip, the operation may take a long time (e.g. to erase > + * flash), so the commands are asynchronous. > + */ > +#define EC_CMD_ADD_ENTROPY 0x0122 > + > +enum add_entropy_action { > + /* Add entropy to the current secret. */ > + ADD_ENTROPY_ASYNC = 0, > + /* > + * Add entropy, and also make sure that the previous secret is erased. > + * (this can be implemented by adding entropy multiple times until > + * all rolback blocks have been overwritten). > + */ > + ADD_ENTROPY_RESET_ASYNC = 1, > + /* Read back result from the previous operation. */ > + ADD_ENTROPY_GET_RESULT = 2, > +}; > + > +struct ec_params_rollback_add_entropy { > + uint8_t action; > +} __ec_align1; > + > +/* > + * Perform a single read of a given ADC channel. > + */ > +#define EC_CMD_ADC_READ 0x0123 > + > +struct ec_params_adc_read { > + uint8_t adc_channel; > +} __ec_align1; > + > +struct ec_response_adc_read { > + int32_t adc_value; > +} __ec_align4; > + > +/* > + * Read back rollback info > + */ > +#define EC_CMD_ROLLBACK_INFO 0x0124 > + > +struct ec_response_rollback_info { > + int32_t id; /* Incrementing number to indicate which region to use. */ > + int32_t rollback_min_version; > + int32_t rw_rollback_version; > +} __ec_align4; > + > + > +/* Issue AP reset */ > +#define EC_CMD_AP_RESET 0x0125 > + > +/*****************************************************************************/ > +/* The command range 0x200-0x2FF is reserved for Rotor. */ > + > +/*****************************************************************************/ > +/* > + * Reserve a range of host commands for the CR51 firmware. > + */ > +#define EC_CMD_CR51_BASE 0x0300 > +#define EC_CMD_CR51_LAST 0x03FF > + > +/*****************************************************************************/ > +/* Fingerprint MCU commands: range 0x0400-0x040x */ > + > +/* Fingerprint SPI sensor passthru command: prototyping ONLY */ > +#define EC_CMD_FP_PASSTHRU 0x0400 > + > +#define EC_FP_FLAG_NOT_COMPLETE 0x1 > + > +struct ec_params_fp_passthru { > + uint16_t len; /* Number of bytes to write then read */ > + uint16_t flags; /* EC_FP_FLAG_xxx */ > + uint8_t data[]; /* Data to send */ > +} __ec_align2; > + > +/* Configure the Fingerprint MCU behavior */ > +#define EC_CMD_FP_MODE 0x0402 > + > +/* Put the sensor in its lowest power mode */ > +#define FP_MODE_DEEPSLEEP (1<<0) > +/* Wait to see a finger on the sensor */ > +#define FP_MODE_FINGER_DOWN (1<<1) > +/* Poll until the finger has left the sensor */ > +#define FP_MODE_FINGER_UP (1<<2) > +/* Capture the current finger image */ > +#define FP_MODE_CAPTURE (1<<3) > +/* Finger enrollment session on-going */ > +#define FP_MODE_ENROLL_SESSION (1<<4) > +/* Enroll the current finger image */ > +#define FP_MODE_ENROLL_IMAGE (1<<5) > +/* Try to match the current finger image */ > +#define FP_MODE_MATCH (1<<6) > +/* Reset and re-initialize the sensor. */ > +#define FP_MODE_RESET_SENSOR (1<<7) > +/* special value: don't change anything just read back current mode */ > +#define FP_MODE_DONT_CHANGE (1<<31) > + > +#define FP_VALID_MODES (FP_MODE_DEEPSLEEP | \ > + FP_MODE_FINGER_DOWN | \ > + FP_MODE_FINGER_UP | \ > + FP_MODE_CAPTURE | \ > + FP_MODE_ENROLL_SESSION | \ > + FP_MODE_ENROLL_IMAGE | \ > + FP_MODE_MATCH | \ > + FP_MODE_RESET_SENSOR | \ > + FP_MODE_DONT_CHANGE) > + > +/* Capture types defined in bits [30..28] */ > +#define FP_MODE_CAPTURE_TYPE_SHIFT 28 > +#define FP_MODE_CAPTURE_TYPE_MASK (0x7 << FP_MODE_CAPTURE_TYPE_SHIFT) > +/* > + * This enum must remain ordered, if you add new values you must ensure that > + * FP_CAPTURE_TYPE_MAX is still the last one. > + */ > +enum fp_capture_type { > + /* Full blown vendor-defined capture (produces 'frame_size' bytes) */ > + FP_CAPTURE_VENDOR_FORMAT = 0, > + /* Simple raw image capture (produces width x height x bpp bits) */ > + FP_CAPTURE_SIMPLE_IMAGE = 1, > + /* Self test pattern (e.g. checkerboard) */ > + FP_CAPTURE_PATTERN0 = 2, > + /* Self test pattern (e.g. inverted checkerboard) */ > + FP_CAPTURE_PATTERN1 = 3, > + /* Capture for Quality test with fixed contrast */ > + FP_CAPTURE_QUALITY_TEST = 4, > + /* Capture for pixel reset value test */ > + FP_CAPTURE_RESET_TEST = 5, > + FP_CAPTURE_TYPE_MAX, > +}; > +/* Extracts the capture type from the sensor 'mode' word */ > +#define FP_CAPTURE_TYPE(mode) (((mode) & FP_MODE_CAPTURE_TYPE_MASK) \ > + >> FP_MODE_CAPTURE_TYPE_SHIFT) > + > +struct ec_params_fp_mode { > + uint32_t mode; /* as defined by FP_MODE_ constants */ > +} __ec_align4; > + > +struct ec_response_fp_mode { > + uint32_t mode; /* as defined by FP_MODE_ constants */ > +} __ec_align4; > + > +/* Retrieve Fingerprint sensor information */ > +#define EC_CMD_FP_INFO 0x0403 > + > +/* Number of dead pixels detected on the last maintenance */ > +#define FP_ERROR_DEAD_PIXELS(errors) ((errors) & 0x3FF) > +/* Unknown number of dead pixels detected on the last maintenance */ > +#define FP_ERROR_DEAD_PIXELS_UNKNOWN (0x3FF) > +/* No interrupt from the sensor */ > +#define FP_ERROR_NO_IRQ (1 << 12) > +/* SPI communication error */ > +#define FP_ERROR_SPI_COMM (1 << 13) > +/* Invalid sensor Hardware ID */ > +#define FP_ERROR_BAD_HWID (1 << 14) > +/* Sensor initialization failed */ > +#define FP_ERROR_INIT_FAIL (1 << 15) > + > +struct ec_response_fp_info_v0 { > + /* Sensor identification */ > + uint32_t vendor_id; > + uint32_t product_id; > + uint32_t model_id; > + uint32_t version; > + /* Image frame characteristics */ > + uint32_t frame_size; > + uint32_t pixel_format; /* using V4L2_PIX_FMT_ */ > + uint16_t width; > + uint16_t height; > + uint16_t bpp; > + uint16_t errors; /* see FP_ERROR_ flags above */ > +} __ec_align4; > + > +struct ec_response_fp_info { > + /* Sensor identification */ > + uint32_t vendor_id; > + uint32_t product_id; > + uint32_t model_id; > + uint32_t version; > + /* Image frame characteristics */ > + uint32_t frame_size; > + uint32_t pixel_format; /* using V4L2_PIX_FMT_ */ > + uint16_t width; > + uint16_t height; > + uint16_t bpp; > + uint16_t errors; /* see FP_ERROR_ flags above */ > + /* Template/finger current information */ > + uint32_t template_size; /* max template size in bytes */ > + uint16_t template_max; /* maximum number of fingers/templates */ > + uint16_t template_valid; /* number of valid fingers/templates */ > + uint32_t template_dirty; /* bitmap of templates with MCU side changes */ > + uint32_t template_version; /* version of the template format */ > +} __ec_align4; > + > +/* Get the last captured finger frame or a template content */ > +#define EC_CMD_FP_FRAME 0x0404 > + > +/* constants defining the 'offset' field which also contains the frame index */ > +#define FP_FRAME_INDEX_SHIFT 28 > +/* Frame buffer where the captured image is stored */ > +#define FP_FRAME_INDEX_RAW_IMAGE 0 > +/* First frame buffer holding a template */ > +#define FP_FRAME_INDEX_TEMPLATE 1 > +#define FP_FRAME_GET_BUFFER_INDEX(offset) ((offset) >> FP_FRAME_INDEX_SHIFT) > +#define FP_FRAME_OFFSET_MASK 0x0FFFFFFF > + > +/* Version of the format of the encrypted templates. */ > +#define FP_TEMPLATE_FORMAT_VERSION 3 > + > +/* Constants for encryption parameters */ > +#define FP_CONTEXT_NONCE_BYTES 12 > +#define FP_CONTEXT_USERID_WORDS (32 / sizeof(uint32_t)) > +#define FP_CONTEXT_TAG_BYTES 16 > +#define FP_CONTEXT_SALT_BYTES 16 > +#define FP_CONTEXT_TPM_BYTES 32 > + > +struct ec_fp_template_encryption_metadata { > + /* > + * Version of the structure format (N=3). > + */ > + uint16_t struct_version; > + /* Reserved bytes, set to 0. */ > + uint16_t reserved; > + /* > + * The salt is *only* ever used for key derivation. The nonce is unique, > + * a different one is used for every message. > + */ > + uint8_t nonce[FP_CONTEXT_NONCE_BYTES]; > + uint8_t salt[FP_CONTEXT_SALT_BYTES]; > + uint8_t tag[FP_CONTEXT_TAG_BYTES]; > +}; > + > +struct ec_params_fp_frame { > + /* > + * The offset contains the template index or FP_FRAME_INDEX_RAW_IMAGE > + * in the high nibble, and the real offset within the frame in > + * FP_FRAME_OFFSET_MASK. > + */ > + uint32_t offset; > + uint32_t size; > +} __ec_align4; > + > +/* Load a template into the MCU */ > +#define EC_CMD_FP_TEMPLATE 0x0405 > + > +/* Flag in the 'size' field indicating that the full template has been sent */ > +#define FP_TEMPLATE_COMMIT 0x80000000 > + > +struct ec_params_fp_template { > + uint32_t offset; > + uint32_t size; > + uint8_t data[]; > +} __ec_align4; > + > +/* Clear the current fingerprint user context and set a new one */ > +#define EC_CMD_FP_CONTEXT 0x0406 > + > +struct ec_params_fp_context { > + uint32_t userid[FP_CONTEXT_USERID_WORDS]; > +} __ec_align4; > + > +#define EC_CMD_FP_STATS 0x0407 > + > +#define FPSTATS_CAPTURE_INV (1 << 0) > +#define FPSTATS_MATCHING_INV (1 << 1) > + > +struct ec_response_fp_stats { > + uint32_t capture_time_us; > + uint32_t matching_time_us; > + uint32_t overall_time_us; > + struct { > + uint32_t lo; > + uint32_t hi; > + } overall_t0; > + uint8_t timestamps_invalid; > + int8_t template_matched; > +} __ec_align2; > + > +#define EC_CMD_FP_SEED 0x0408 > +struct ec_params_fp_seed { > + /* > + * Version of the structure format (N=3). > + */ > + uint16_t struct_version; > + /* Reserved bytes, set to 0. */ > + uint16_t reserved; > + /* Seed from the TPM. */ > + uint8_t seed[FP_CONTEXT_TPM_BYTES]; > +} __ec_align4; > + > +/*****************************************************************************/ > +/* Touchpad MCU commands: range 0x0500-0x05FF */ > + > +/* Perform touchpad self test */ > +#define EC_CMD_TP_SELF_TEST 0x0500 > + > +/* Get number of frame types, and the size of each type */ > +#define EC_CMD_TP_FRAME_INFO 0x0501 > + > +struct ec_response_tp_frame_info { > + uint32_t n_frames; > + uint32_t frame_sizes[0]; > +} __ec_align4; > + > +/* Create a snapshot of current frame readings */ > +#define EC_CMD_TP_FRAME_SNAPSHOT 0x0502 > + > +/* Read the frame */ > +#define EC_CMD_TP_FRAME_GET 0x0503 > + > +struct ec_params_tp_frame_get { > + uint32_t frame_index; > + uint32_t offset; > + uint32_t size; > +} __ec_align4; > + > +/*****************************************************************************/ > +/* EC-EC communication commands: range 0x0600-0x06FF */ > + > +#define EC_COMM_TEXT_MAX 8 > + > +/* > + * Get battery static information, i.e. information that never changes, or > + * very infrequently. > + */ > +#define EC_CMD_BATTERY_GET_STATIC 0x0600 > + > +/** > + * struct ec_params_battery_static_info - Battery static info parameters > + * @index: Battery index. > + */ > +struct ec_params_battery_static_info { > + uint8_t index; > +} __ec_align_size1; > + > +/** > + * struct ec_response_battery_static_info - Battery static info response > + * @design_capacity: Battery Design Capacity (mAh) > + * @design_voltage: Battery Design Voltage (mV) > + * @manufacturer: Battery Manufacturer String > + * @model: Battery Model Number String > + * @serial: Battery Serial Number String > + * @type: Battery Type String > + * @cycle_count: Battery Cycle Count > + */ > +struct ec_response_battery_static_info { > + uint16_t design_capacity; > + uint16_t design_voltage; > + char manufacturer[EC_COMM_TEXT_MAX]; > + char model[EC_COMM_TEXT_MAX]; > + char serial[EC_COMM_TEXT_MAX]; > + char type[EC_COMM_TEXT_MAX]; > + /* TODO(crbug.com/795991): Consider moving to dynamic structure. */ > + uint32_t cycle_count; > +} __ec_align4; > + > +/* > + * Get battery dynamic information, i.e. information that is likely to change > + * every time it is read. > + */ > +#define EC_CMD_BATTERY_GET_DYNAMIC 0x0601 > + > +/** > + * struct ec_params_battery_dynamic_info - Battery dynamic info parameters > + * @index: Battery index. > + */ > +struct ec_params_battery_dynamic_info { > + uint8_t index; > +} __ec_align_size1; > + > +/** > + * struct ec_response_battery_dynamic_info - Battery dynamic info response > + * @actual_voltage: Battery voltage (mV) > + * @actual_current: Battery current (mA); negative=discharging > + * @remaining_capacity: Remaining capacity (mAh) > + * @full_capacity: Capacity (mAh, might change occasionally) > + * @flags: Flags, see EC_BATT_FLAG_* > + * @desired_voltage: Charging voltage desired by battery (mV) > + * @desired_current: Charging current desired by battery (mA) > + */ > +struct ec_response_battery_dynamic_info { > + int16_t actual_voltage; > + int16_t actual_current; > + int16_t remaining_capacity; > + int16_t full_capacity; > + int16_t flags; > + int16_t desired_voltage; > + int16_t desired_current; > +} __ec_align2; > + > +/* > + * Control charger chip. Used to control charger chip on the slave. > + */ > +#define EC_CMD_CHARGER_CONTROL 0x0602 > + > +/** > + * struct ec_params_charger_control - Charger control parameters > + * @max_current: Charger current (mA). Positive to allow base to draw up to > + * max_current and (possibly) charge battery, negative to request current > + * from base (OTG). > + * @otg_voltage: Voltage (mV) to use in OTG mode, ignored if max_current is > + * >= 0. > + * @allow_charging: Allow base battery charging (only makes sense if > + * max_current > 0). > + */ > +struct ec_params_charger_control { > + int16_t max_current; > + uint16_t otg_voltage; > + uint8_t allow_charging; > +} __ec_align_size1; > + > +/*****************************************************************************/ > +/* > + * Reserve a range of host commands for board-specific, experimental, or > + * special purpose features. These can be (re)used without updating this file. > + * > + * CAUTION: Don't go nuts with this. Shipping products should document ALL > + * their EC commands for easier development, testing, debugging, and support. > + * > + * All commands MUST be #defined to be 4-digit UPPER CASE hex values > + * (e.g., 0x00AB, not 0xab) for CONFIG_HOSTCMD_SECTION_SORTED to work. > + * > + * In your experimental code, you may want to do something like this: > + * > + * #define EC_CMD_MAGIC_FOO 0x0000 > + * #define EC_CMD_MAGIC_BAR 0x0001 > + * #define EC_CMD_MAGIC_HEY 0x0002 > + * > + * DECLARE_PRIVATE_HOST_COMMAND(EC_CMD_MAGIC_FOO, magic_foo_handler, > + * EC_VER_MASK(0); > + * > + * DECLARE_PRIVATE_HOST_COMMAND(EC_CMD_MAGIC_BAR, magic_bar_handler, > + * EC_VER_MASK(0); > + * > + * DECLARE_PRIVATE_HOST_COMMAND(EC_CMD_MAGIC_HEY, magic_hey_handler, > + * EC_VER_MASK(0); > + */ > +#define EC_CMD_BOARD_SPECIFIC_BASE 0x3E00 > +#define EC_CMD_BOARD_SPECIFIC_LAST 0x3FFF > + > +/* > + * Given the private host command offset, calculate the true private host > + * command value. > + */ > +#define EC_PRIVATE_HOST_COMMAND_VALUE(command) \ > + (EC_CMD_BOARD_SPECIFIC_BASE + (command)) > > /*****************************************************************************/ > /* > @@ -3457,4 +5694,10 @@ struct ec_response_usb_pd_mux_info { > #define EC_LPC_ADDR_OLD_PARAM EC_HOST_CMD_REGION1 > #define EC_OLD_PARAM_SIZE EC_HOST_CMD_REGION_SIZE > > +#endif /* !__ACPI__ */ > + > +#ifdef __cplusplus > +} > +#endif > + > #endif /* __CROS_EC_COMMANDS_H */

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Re: [alsa-devel] [PATCH 3/3] mfd: cros: Update EC to use BIT() and merge
by Jonathan Cameron 30 Mar '19

30 Mar '19

On Fri, 29 Mar 2019 10:56:37 -0700 Gwendal Grignou <gwendal(a)chromium.org> wrote: What is "and merge" referring to in the title? > Instead of using 1 << ..., use BIT() cleaner and safer. > Fix minor fixes since major cros_ec_commands.h cleanup was uploaded. > > TEST=compile. Personally I'd prefer this is a a couple of patches as several different changes going on. A few minor comments below. > > Signed-off-by: Gwendal Grignou <gwendal(a)chromium.org> > --- > include/linux/mfd/cros_ec_commands.h | 268 ++++++++++++++------------- > 1 file changed, 143 insertions(+), 125 deletions(-) > > diff --git a/include/linux/mfd/cros_ec_commands.h > b/include/linux/mfd/cros_ec_commands.h > index ea25e1644d23..5c4764e3db6f 100644 > --- a/include/linux/mfd/cros_ec_commands.h > +++ b/include/linux/mfd/cros_ec_commands.h > @@ -46,7 +46,7 @@ extern "C"{ > #define EC_PROTO_VERSION 0x00000002 ... > > /** > * struct ec_response_get_protocol_info - Response to the get protocol info. > @@ -1347,12 +1347,12 @@ enum ec_feature_code { > EC_FEATURE_AUDIO_CODEC = 38, > /* EC Supports SCP. */ > EC_FEATURE_SCP = 39, > - /* The MCU is Intel Integrated Sensor Hub */ > + /* The MCU is an Integrated Sensor Hub */ Unrelated from the other changes so I'd like to see it in a separate patch. > EC_FEATURE_ISH = 40, > }; ... > enum ec_led_colors { > EC_LED_COLOR_RED = 0, > @@ -2438,6 +2438,7 @@ enum motionsensor_chip { > MOTIONSENSE_CHIP_LIS2DE = 15, > MOTIONSENSE_CHIP_LIS2MDL = 16, > MOTIONSENSE_CHIP_LSM6DS3 = 17, > + MOTIONSENSE_CHIP_LSM6DSO = 18, Why is this here? > MOTIONSENSE_CHIP_MAX, > }; > ... > > enum mkbp_config_valid { Bit odd that this was an enum in the first place.. I checked and it doesn't seem that the type is used anywhere, so I would just make this set of defines. > - EC_MKBP_VALID_SCAN_PERIOD = 1 << 0, > - EC_MKBP_VALID_POLL_TIMEOUT = 1 << 1, > - EC_MKBP_VALID_MIN_POST_SCAN_DELAY = 1 << 3, > - EC_MKBP_VALID_OUTPUT_SETTLE = 1 << 4, > - EC_MKBP_VALID_DEBOUNCE_DOWN = 1 << 5, > - EC_MKBP_VALID_DEBOUNCE_UP = 1 << 6, > - EC_MKBP_VALID_FIFO_MAX_DEPTH = 1 << 7, > + EC_MKBP_VALID_SCAN_PERIOD = BIT(0), > + EC_MKBP_VALID_POLL_TIMEOUT = BIT(1), > + EC_MKBP_VALID_MIN_POST_SCAN_DELAY = BIT(3), > + EC_MKBP_VALID_OUTPUT_SETTLE = BIT(4), > + EC_MKBP_VALID_DEBOUNCE_DOWN = BIT(5), > + EC_MKBP_VALID_DEBOUNCE_UP = BIT(6), > + EC_MKBP_VALID_FIFO_MAX_DEPTH = BIT(7), > }; > ...

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[alsa-devel] 1.18: alsa-lib build fails with LTO
by GitHub issues - opened 30 Mar '19

30 Mar '19

alsa-project/alsa-lib issue #6 was opened from kloczek: <pre>make[2]: Entering directory '/home/tkloczko/rpmbuild/BUILD/alsa-lib-1.1.8/src' /bin/sh ../libtool --tag=CC --mode=link gcc -O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -Wp,-D_GLIBCXX_ASSERTIONS -fexceptions -fstack-protector-strong -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -mtune=generic -fasynchronous-unwind-tables -fstack-clash-protection -fcf-protection -flto -version-info 2:0:0 -Wl,--version-script=Versions -Wl,--no-undefined -Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -flto -fuse-linker-plugin -o libasound.la -rpath /usr/lib64 conf.lo confmisc.lo input.lo output.lo async.lo error.lo dlmisc.lo socket.lo shmarea.lo userfile.lo names.lo control/libcontrol.la mixer/libmixer.la pcm/libpcm.la timer/libtimer.la rawmidi/librawmidi.la hwdep/libhwdep.la seq/libseq.la ucm/libucm.la topology/libtopology.la -lm -ldl -lpthread -lrt libtool: link: gcc -shared -fPIC -DPIC .libs/conf.o .libs/confmisc.o .libs/input.o .libs/output.o .libs/async.o .libs/error.o .libs/dlmisc.o .libs/socket.o .libs/shmarea.o .libs/userfile.o .libs/names.o -Wl,--whole-archive control/.libs/libcontrol.a mixer/.libs/libmixer.a pcm/.libs/libpcm.a timer/.libs/libtimer.a rawmidi/.libs/librawmidi.a hwdep/.libs/libhwdep.a seq/.libs/libseq.a ucm/.libs/libucm.a topology/.libs/libtopology.a -Wl,--no-whole-archive -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -O2 -g -fstack-protector-strong -grecord-gcc-switches -m64 -mtune=generic -flto -Wl,--version-script=Versions -Wl,--no-undefined -Wl,-z -Wl,relro -Wl,--as-needed -Wl,-z -Wl,now -flto -fuse-linker-plugin -lm -ldl -lpthread -lrt -Wl,-soname -Wl,libasound.so.2 -o .libs/libasound.so.2.0.0 /tmp/ccROQp1D.s: Assembler messages: /tmp/ccROQp1D.s: Error: invalid attempt to declare external version name as default in symbol `snd_dlopen@@ALSA_1.1.6' lto-wrapper: fatal error: gcc returned 1 exit status compilation terminated. /usr/bin/ld: error: lto-wrapper failed </pre> Issue URL : https://github.com/alsa-project/alsa-lib/issues/6 Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [BACKPORT 4.4.y 00/25] candidates from spreadtrum 4.4 product kernel
by Arnd Bergmann 30 Mar '19

30 Mar '19

I took a scripted approach to look at some product kernels for patches backported into vendor kernels. This is a set of (mostly) bugfixes I found in Spreadtrum's linux-4.4 kernel that are missing in 4.4.176: ffedbd2210f2 mmc: pwrseq: constify mmc_pwrseq_ops structures c10368897e10 ALSA: compress: add support for 32bit calls in a 64bit kernel 64a67d4762ce mmc: pwrseq_simple: Make reset-gpios optional to match doc 4ec0ef3a8212 USB: iowarrior: fix oops with malicious USB descriptors e5905ff1281f mmc: debugfs: Add a restriction to mmc debugfs clock setting 4ec96b4cbde8 mmc: make MAN_BKOPS_EN message a debug ed9feec72fc1 mmc: sanitize 'bus width' in debug output 10a16a01d8f7 mmc: core: shut up "voltage-ranges unspecified" pr_info() 9772b47a4c29 usb: dwc3: gadget: Fix suspend/resume during device mode 6afedcd23cfd arm64: mm: Add trace_irqflags annotations to do_debug_exception() 437db4c6e798 mmc: mmc: Attempt to flush cache before reset e51534c80660 mmc: core: fix using wrong io voltage if mmc_select_hs200 fails e4c5800a3991 mm/rmap: replace BUG_ON(anon_vma->degree) with VM_WARN_ON 04c080080855 extcon: usb-gpio: Don't miss event during suspend/resume 78283edf2c01 kbuild: setlocalversion: print error to STDERR c526c62d565e usb: gadget: composite: fix dereference after null check coverify warning 511a36d2f357 usb: gadget: Add the gserial port checking in gs_start_tx() 1712c9373f98 mmc: core: don't try to switch block size for dual rate mode 5ea8ea2cb7f1 tcp/dccp: drop SYN packets if accept queue is full e1dc9b08051a serial: sprd: adjust TIMEOUT to a big value 81be24d263db Hang/soft lockup in d_invalidate with simultaneous calls 6f44a0bacb79 arm64: traps: disable irq in die() b7d44c36a6f6 usb: renesas_usbhs: gadget: fix unused-but-set-variable warning 4350782570b9 serial: sprd: clear timeout interrupt only rather than all interrupts 3f3295709ede lib/int_sqrt: optimize small argument 32fd87b3bbf5 USB: core: only clean up what we allocated Al Viro (1): Hang/soft lockup in d_invalidate with simultaneous calls Andrey Konovalov (1): USB: core: only clean up what we allocated Baolin Wang (1): usb: gadget: Add the gserial port checking in gs_start_tx() Chuanxiao Dong (1): mmc: debugfs: Add a restriction to mmc debugfs clock setting Dong Aisheng (1): mmc: core: fix using wrong io voltage if mmc_select_hs200 fails Eric Dumazet (1): tcp/dccp: drop SYN packets if accept queue is full James Morse (1): arm64: mm: Add trace_irqflags annotations to do_debug_exception() Josh Boyer (1): USB: iowarrior: fix oops with malicious USB descriptors Julia Lawall (1): mmc: pwrseq: constify mmc_pwrseq_ops structures Konstantin Khlebnikov (1): mm/rmap: replace BUG_ON(anon_vma->degree) with VM_WARN_ON Lanqing Liu (1): serial: sprd: clear timeout interrupt only rather than all interrupts Martin Fuzzey (1): mmc: pwrseq_simple: Make reset-gpios optional to match doc Peter Chen (1): usb: gadget: composite: fix dereference after null check coverify warning Peter Zijlstra (1): lib/int_sqrt: optimize small argument Qiao Zhou (1): arm64: traps: disable irq in die() Ravindra Lokhande (1): ALSA: compress: add support for 32bit calls in a 64bit kernel Roger Quadros (2): usb: dwc3: gadget: Fix suspend/resume during device mode extcon: usb-gpio: Don't miss event during suspend/resume Russell King (1): mmc: core: shut up "voltage-ranges unspecified" pr_info() Wei Qiao (1): serial: sprd: adjust TIMEOUT to a big value Wolfram Sang (3): mmc: make MAN_BKOPS_EN message a debug mmc: sanitize 'bus width' in debug output kbuild: setlocalversion: print error to STDERR Yoshihiro Shimoda (1): usb: renesas_usbhs: gadget: fix unused-but-set-variable warning Ziyuan Xu (1): mmc: core: don't try to switch block size for dual rate mode arch/arm64/kernel/traps.c | 8 +++++-- arch/arm64/mm/fault.c | 33 ++++++++++++++++++-------- drivers/extcon/extcon-usb-gpio.c | 3 +++ drivers/mmc/core/core.c | 13 ++++++---- drivers/mmc/core/debugfs.c | 2 +- drivers/mmc/core/mmc.c | 16 +++++++++---- drivers/mmc/core/pwrseq.h | 2 +- drivers/mmc/core/pwrseq_emmc.c | 2 +- drivers/mmc/core/pwrseq_simple.c | 24 ++++++++++++------- drivers/tty/serial/sprd_serial.c | 6 +++-- drivers/usb/core/config.c | 9 ++++--- drivers/usb/dwc3/gadget.c | 6 +++++ drivers/usb/gadget/composite.c | 2 ++ drivers/usb/gadget/function/u_serial.c | 7 +++++- drivers/usb/misc/iowarrior.c | 6 +++++ drivers/usb/renesas_usbhs/mod_gadget.c | 5 +--- fs/dcache.c | 10 ++++---- include/net/inet_connection_sock.h | 5 ---- lib/int_sqrt.c | 3 +++ mm/rmap.c | 2 +- net/dccp/ipv4.c | 8 +------ net/dccp/ipv6.c | 2 +- net/ipv4/tcp_input.c | 8 +------ scripts/setlocalversion | 2 +- sound/core/compress_offload.c | 13 ++++++++++ 25 files changed, 126 insertions(+), 71 deletions(-) -- 2.20.0 This is the full list of patches that were backported and are not in 4.4.y, but as usual most of them did not appear to make sense for stable kernels. 100 33 da5ce874f8ca f2fs: release locks before return in f2fs_ioc_gc_range() 100 100 1dc0f8991d4d f2fs: fix to avoid race in between atomic write and background GC 100 27 b27bc8091ccf f2fs: do gc in greedy mode for whole range if gc_urgent mode is set 100 100 782911f491e7 f2fs: set readdir_ra by default 33 33 81286d3e31b7 staging: android: ion: Remove check of idev->debug_root 100 80 ae6650163c66 loop: fix concurrent lo_open/lo_release 100 100 466a2b42d676 cpufreq: schedutil: Use idle_calls counter of the remote CPU 100 100 32fd87b3bbf5 USB: core: only clean up what we allocated 100 100 3f3295709ede lib/int_sqrt: optimize small argument 50 30 e22cdc3fc599 sched/isolcpus: Fix "isolcpus=" boot parameter handling when !CONFIG_CPUMASK_OFFSTACK 100 85 0abd8e70d24b f2fs: clear radix tree dirty tag of pages whose dirty flag is cleared 100 100 84a23fbe96b4 f2fs: clear FI_HOT_DATA correctly 91 91 12ac1d0f6c3e genirq: Make sparse_irq_lock protect what it should protect 100 55 c49cbc19b31e cpufreq: schedutil: Always process remote callback with slow switching 100 100 e2cabe48c20e cpufreq: schedutil: Don't restrict kthread to related_cpus unnecessarily 78 78 99d14d0e16fa cpufreq: Process remote callbacks from any CPU if the platform permits 31 61 674e75411fc2 sched: cpufreq: Allow remote cpufreq callbacks 50 100 4350782570b9 serial: sprd: clear timeout interrupt only rather than all interrupts 100 100 b7d44c36a6f6 usb: renesas_usbhs: gadget: fix unused-but-set-variable warning 100 100 6f44a0bacb79 arm64: traps: disable irq in die() 100 33 04dfc23006a2 f2fs: show more info if fail to issue discard 71 28 e41e6d75e501 f2fs: split wio_mutex 100 100 773a9ef85f02 mmc: pwrseq: Add reset callback to the struct mmc_pwrseq_ops 100 100 81be24d263db Hang/soft lockup in d_invalidate with simultaneous calls 100 100 e9256e142f59 mmc: pwrseq_simple: Parse DTS for the power-off-delay-us property 100 100 e1dc9b08051a serial: sprd: adjust TIMEOUT to a big value 100 100 6c3acd97572b f2fs: allocate hot_data for atomic writes 75 100 bdd154436077 USB: serial: spcp8x5: simplify endpoint check 45 100 590298b22325 USB: serial: pl2303: simplify endpoint check 68 100 32814c87f446 USB: serial: oti6858: simplify endpoint check 85 100 8ee1592d125a USB: serial: omninet: simplify endpoint check 42 100 206ff831bebb USB: serial: mos7720: simplify endpoint check 84 100 35194572b4ed USB: serial: kobil_sct: simplify endpoint check 75 100 b714d5dc0631 USB: serial: keyspan_pda: simplify endpoint check 69 100 fb527736ebcc USB: serial: iuu_phoenix: simplify endpoint check 65 100 e7d6507e5ba7 USB: serial: digi_acceleport: simplify endpoint check 81 100 d183b9b43390 USB: serial: cyberjack: simplify endpoint check 60 100 fe190ed0d602 xhci: Do not halt the host until both HCD have disconnected their devices. 100 100 f759741d9d91 block: Fix oops in locked_inode_to_wb_and_lock_list() 100 100 773dc118756b mmc: core: Fix access to HS400-ES devices 64 64 e93b9865251a f2fs: add ovp valid_blocks check for bg gc victim to fg_gc 30 49 942fd3192f83 f2fs: check last page index in cached bio to decide submission 71 100 bcb7440e76a9 extcon: usb-gpio: Add pinctrl operation during system PM 66 100 5278204c9818 xhci: use the trb_to_noop() helper for command trbs 100 100 bc88c10d7e69 locking/spinlock/debug: Remove spinlock lockup detection code 100 100 d40a43af0a57 f2fs: fix an infinite loop when flush nodes in cp 89 86 541332a13b1d extcon: usb-gpio: Add VBUS detection support 100 50 65aca3205046 usb: dwc3: gadget: clear events in top-half handler 77 100 ebbb2d59398f usb: dwc3: gadget: use evt->cache for processing events 66 100 d9fa4c63f766 usb: dwc3: core: add a event buffer cache 100 100 9ad587710a2f usb: gadget: composite: remove unnecessary & operation 100 76 ef3d232245ab mmc: mmc: Relax checking for switch errors after HS200 switch 100 100 e173f8911f09 mmc: core: Update CMD13 polling policy when switch to HS DDR mode 68 63 aa33ce3c411a mmc: core: Enable __mmc_switch() to change bus speed timing for the host 100 33 5ec32f84111a mmc: core: Check SWITCH_ERROR bit from each CMD13 response when polling 50 50 625228fa3e01 mmc: core: Rename ignore_crc to retry_crc_err to reflect its purpose 100 100 89e57aedda33 mmc: core: Remove redundant __mmc_send_status() 33 33 437590a123b6 mmc: core: Retry instead of ignore at CRC errors when polling for busy 100 100 c2c24819b280 mmc: core: Don't power off the card when starting the host 55 50 716bdb8953c7 mmc: core: Factor out code related to polling in __mmc_switch() 23 89 cb26ce069ffa mmc: core: Clarify code which deals with polling in __mmc_switch() 100 100 5ea8ea2cb7f1 tcp/dccp: drop SYN packets if accept queue is full 72 100 8e5bfa8c1f84 sched/autogroup: Do not use autogroup->tg in zombie threads 100 100 8fdd136f2200 cfg80211: add bitrate for 20MHz MCS 9 75 100 fd9afd3cbe40 usb: gadget: u_ether: remove interrupt throttling 100 100 fe1b5700c70f mmc: mmc: Use 500ms as the default generic CMD6 timeout 100 100 1720d3545b77 mmc: core: switch to 1V8 or 1V2 for hs400es mode 100 100 e932835377f9 f2fs: check return value of write_checkpoint during fstrim 100 100 1712c9373f98 mmc: core: don't try to switch block size for dual rate mode 63 81 721e0497172f mmc: pwrseq-simple: Add an optional post-power-on-delay 100 100 00af62330c39 usb: dwc3: core: Move the mode setting to the right place 100 75 b1149ad917b7 coresight: always use stashed trace id value in etm4_trace_id 80 80 a399d233078e sched/core: Fix incorrect utilization accounting when switching to fair class 100 60 9d7aba7786b6 Revert "usb: dwc3: gadget: always decrement by 1" 100 100 511a36d2f357 usb: gadget: Add the gserial port checking in gs_start_tx() 100 100 c526c62d565e usb: gadget: composite: fix dereference after null check coverify warning 100 100 78283edf2c01 kbuild: setlocalversion: print error to STDERR 100 100 bb4eecf23be2 mmc: Change the max discard sectors and erase response when HW busy detect 100 100 6ae3e537eab9 mmc: core: expose MMC_CAP2_NO_* to dt 100 100 5f1d1434b7a0 Documentation: mmc: add description for new no-sd* and no-mmc 100 100 a0c3b68c72a3 mmc: core: Allow hosts to specify non-support for MMC commands 100 100 1b8d79c54944 mmc: core: Allow hosts to specify non-support for SD commands 100 100 649c6059d237 mmc: mmc: Fix HS switch failure in mmc_select_hs400() 61 100 08573eaf1a70 mmc: mmc: do not use CMD13 to get status after speed mode switch 33 100 bc26235bbd79 mmc: debugfs: add HS400 enhanced strobe description 74 100 81ac2af65793 mmc: core: implement enhanced strobe support 100 100 ef29c0e273b8 mmc: core: add mmc-hs400-enhanced-strobe support 100 100 a60119ce9434 Documentation: mmc: add mmc-hs400-enhanced-strobe 99 78 48b4800a1c6a zsmalloc: page migration support 88 79 bfd093f5e7f0 zsmalloc: use freeobj for index 48 62 4aa409cab7c3 zsmalloc: separate free_zspage from putback_zspage 66 66 3783689a1aa8 zsmalloc: introduce zspage structure 62 86 bdb0af7ca8f0 zsmalloc: factor page chain functionality out 100 100 1b8320b620d6 zsmalloc: use bit_spin_lock 50 68 1fc6e27d7b86 zsmalloc: keep max_object in size_class 100 85 b1123ea6d3b3 mm: balloon: use general non-lru movable page feature 88 93 bda807d44454 mm: migrate: support non-lru movable page migration 90 90 c6c919eb90e0 mm: use put_page() to free page instead of putback_lru_page() 92 100 72704f876f50 dwc3: gadget: Implement the suspend entry event handler 100 100 da1410be21bf usb: dwc3: gadget: Add the suspend state checking when stopping gadget 33 100 13fa2e69b1dd usb: dwc3: gadget: disable XFER_NOT_READY 92 35 361572b5f7a9 usb: dwc3: gadget: Handle TRB index 0 when full or empty 100 100 7d0a038b130c usb: dwc3: gadget: Account for link TRB in TRBs left 100 100 89bc856e5a74 usb: dwc3: gadget: Don't prepare TRBs if no space 75 100 0d25744ad107 usb: dwc3: gadget: Initialize the TRB ring 79 86 fc8bb91bc83e usb: dwc3: implement runtime PM 77 100 4cb4221764ef usb: dwc3: gadget: fix for possible endpoint disable race 45 29 51f5d49ad6f0 usb: dwc3: core: simplify suspend/resume operations 79 91 c499ff71ff2a usb: dwc3: core: re-factor init and exit paths 55 100 bcdb3272e889 usb: dwc3: core: move fladj to dwc3 structure 68 80 c4233573f6ee usb: dwc3: gadget: prepare TRBs on update transfers too 100 100 7f370ed0cfe9 usb: dwc3: core: get rid of DWC3_PM_OPS macro 73 91 9f8a67b65a49 usb: dwc3: gadget: fix gadget suspend/resume 68 93 d7be295243bb usb: dwc3: gadget: re-factor ->udc_start and ->udc_stop 100 100 058b6659e98f extcon: usb-gpio: add device binding for platform device 100 100 04c080080855 extcon: usb-gpio: Don't miss event during suspend/resume 100 100 5fc363232ae7 uas: remove can_queue set in host template 100 100 975756c41332 f2fs: avoid ENOSPC fault in the recovery process 80 40 c41f3cc3ae34 f2fs: inject page allocation failures 43 26 da011cc0da8c f2fs: move node pages only in victim section during GC 100 100 1ee4716585ed zsmalloc: remove unused pool param in obj_free 33 66 830e4bc5baa9 zsmalloc: clean up many BUG_ON 67 71 36b68aae8e39 usb: dwc3: gadget: use link TRB for all endpoint types 62 62 c28f82595dde usb: dwc3: switch trb enqueue/dequeue and first_trb_index to u8 100 100 e4c5800a3991 mm/rmap: replace BUG_ON(anon_vma->degree) with VM_WARN_ON 100 100 e51534c80660 mmc: core: fix using wrong io voltage if mmc_select_hs200 fails 100 100 437db4c6e798 mmc: mmc: Attempt to flush cache before reset 100 100 87e88659afd1 mmc: core: drop unnecessary bit checking 97 96 d97a1e5d7cd2 mmc: pwrseq: convert to proper platform device 75 100 f01b72d0fd53 mmc: pwrseq_emmc: add to_pwrseq_emmc() macro 85 100 5b96fea730ab mmc: pwrseq_simple: add to_pwrseq_simple() macro 100 100 4e6c71788d6b mmc: core: Do regular power cycle when lacking eMMC HW reset support 100 100 6afedcd23cfd arm64: mm: Add trace_irqflags annotations to do_debug_exception() 33 100 9772b47a4c29 usb: dwc3: gadget: Fix suspend/resume during device mode 100 100 a0747eb81c1d mmc: core: remove redundant memset of sdio_read_cccr 100 100 0076c71e37cc mmc: core: remove redundant memset of mmc_decode_cid 100 100 07d97d872359 mmc: core: report tuning command execution failure reason 100 77 cf925747d20b mmc: core: improve mmc_of_parse_voltage() to return better status 75 100 10a16a01d8f7 mmc: core: shut up "voltage-ranges unspecified" pr_info() 100 100 ed9feec72fc1 mmc: sanitize 'bus width' in debug output 100 77 6067bafe44d7 mmc: core: use the defined function to check whether card is removable 100 100 4ec96b4cbde8 mmc: make MAN_BKOPS_EN message a debug 100 100 e5905ff1281f mmc: debugfs: Add a restriction to mmc debugfs clock setting 100 100 0899e7419387 mmc: remove unnecessary assignment statements before return 100 100 62c03ca3ffa1 mmc: core: pwrseq_simple: remove unused header file 33 33 85ead8185a76 f2fs: delete unnecessary wait for page writeback 100 100 4ec0ef3a8212 USB: iowarrior: fix oops with malicious USB descriptors 100 100 5821a33b9bbd Staging: Android: align code with open parenthesis in ion_carveout_heap.c 100 100 9f93a8a0ba91 crypto: api - Introduce crypto_queue_len() helper function 100 92 06b241f32c71 mm: __delete_from_page_cache show Bad page if mapped 100 100 c0992d0f5484 USB: serial: option: add support for Quectel UC20 71 42 3158a8d416f4 USB: option: add support for SIM7100E 87 50 ff4e2494dc17 USB: serial: option: Adding support for Telit LE922 90 100 64a67d4762ce mmc: pwrseq_simple: Make reset-gpios optional to match doc 100 100 c10368897e10 ALSA: compress: add support for 32bit calls in a 64bit kernel 99 100 a5beaaf39455 usb: gadget: Add the console support for usb-to-serial port 100 100 100a606d54a0 mmc: core: Introduce MMC_CAP2_NO_SDIO cap 66 100 ffedbd2210f2 mmc: pwrseq: constify mmc_pwrseq_ops structures 66 100 1ff2575bcf42 mmc: core: Check for non-removable cards earlier in the error path 100 100 c29536e85b5f mmc: core: Make runtime resume default behavior for MMC/SD 100 100 d234d2123fa7 mmc: core: Keep host claimed in mmc_rescan() while calling host ops 100 100 86236813ff23 mmc: core: Invoke ->card_event() callback only when needed Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-fsdevel(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-mm(a)kvack.org Cc: dccp(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org

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[alsa-devel] [PATCH v13 0/6] Media Device Allocator API
by Shuah Khan 29 Mar '19

29 Mar '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. The primary focus for testing The patch series is making sure media device is released when both drivers release the media device with a series of unbind/binds on both drivers. - both au0828 and snd-usb-aduio as built-in - both au0828 and snd-usb-aduio as modules - au0828 as module and snd-usb-aduio as built-in - au0828 as built-in and snd-usb-aduio as module Test results can be found at: https://docs.google.com/document/d/1RMF8Rwj7xHJEoOx6_K2f-REgZJ63BMeAVEf-CV-… Changes since v12: - Patch 1: Fixed prototype warns from media_dev_allocator.c. Removed dependency on find_module() by adding struct module to input args. Still need module name to pass into media_device API. - Patch 2 & 4: Update media dev allocator api calls to pass in struct module pointer. - No changes to Patches 3 & 5. - Added patch 6 with a test. It can go in separately. Changes since v11: - Patch 1: Add CONFIG_MODULES dependency in media_dev_allocator files. to fix compile errors when CONFIG_MODULES is disabled. - Patch 2, 3: No changes. - Patch 4: Fix sparse error reported by Dan Carpenter. - Patch 5: Fix warns found by Hans Verkuil. - v11 was tested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. The test plan used for testing resource sharing could serve as a regression test plan and the test results can be found at: - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (6): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies selftests: media_dev_allocator api test Documentation/media/kapi/mc-core.rst | 41 +++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 133 +++++++ drivers/media/usb/au0828/Kconfig | 2 + drivers/media/usb/au0828/au0828-core.c | 195 ++++++++--- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 63 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 327 ++++++++++++++++++ sound/usb/media.h | 74 ++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + .../media_tests/media_dev_allocator.sh | 81 +++++ 20 files changed, 956 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h create mode 100755 tools/testing/selftests/media_tests/media_dev_allocator.sh -- 2.17.1

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[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3
by Shunli Wang 29 Mar '19

29 Mar '19

ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 Shunli Wang (1): ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 40 +++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: hdmi: implement component supsend/resume callback
by libin.yang＠intel.com 28 Mar '19

28 Mar '19

From: Libin Yang <libin.yang(a)intel.com> Implement the hdmi codec component driver's suspend/resume callback. This can make sure that the dapm event is triggered after the display audio power domain is turned on if bus_control is set. Also this patch removes the codec device PM suspend/resume. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 51 +++++++++++++++++++++++--------------------- 1 file changed, 27 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..478c6f2 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1873,36 +1873,27 @@ static void hdmi_codec_remove(struct snd_soc_component *component) pm_runtime_disable(&hdev->dev); } -#ifdef CONFIG_PM_SLEEP -static int hdmi_codec_resume(struct device *dev) +static int hdmi_component_suspend(struct snd_soc_component *component) { - struct hdac_device *hdev = dev_to_hdac_dev(dev); - struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); - int ret; + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; - ret = pm_runtime_force_resume(dev); - if (ret < 0) - return ret; - /* - * As the ELD notify callback request is not entertained while the - * device is in suspend state. Need to manually check detection of - * all pins here. pin capablity change is not support, so use the - * already set pin caps. - * - * NOTE: this is safe to call even if the codec doesn't actually resume. - * The pin check involves only with DRM audio component hooks, so it - * works even if the HD-audio side is still dreaming peacefully. - */ - hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); - return 0; + return pm_runtime_force_suspend(&hdev->dev); +} + +static int hdmi_component_resume(struct snd_soc_component *component) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + + return pm_runtime_force_resume(&hdev->dev); } -#else -#define hdmi_codec_resume NULL -#endif static const struct snd_soc_component_driver hdmi_hda_codec = { .probe = hdmi_codec_probe, .remove = hdmi_codec_remove, + .suspend = hdmi_component_suspend, + .resume = hdmi_component_resume, .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, @@ -2104,6 +2095,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_link *hlink = NULL; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); dev_dbg(dev, "Enter: %s\n", __func__); @@ -2128,6 +2120,18 @@ static int hdac_hdmi_runtime_resume(struct device *dev) snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* + * As the ELD notify callback request is not entertained while the + * device is in suspend state. Need to manually check detection of + * all pins here. pin capablity change is not support, so use the + * already set pin caps. + * + * NOTE: this is safe to call even if the codec doesn't actually resume. + * The pin check involves only with DRM audio component hooks, so it + * works even if the HD-audio side is still dreaming peacefully. + */ + hdac_hdmi_present_sense_all_pins(hdev, hdmi, false); + return 0; } #else @@ -2137,7 +2141,6 @@ static int hdac_hdmi_runtime_resume(struct device *dev) static const struct dev_pm_ops hdac_hdmi_pm = { SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, hdmi_codec_resume) }; static const struct hda_device_id hdmi_list[] = { -- 2.7.4

3 3

[alsa-devel] [PATCH V2] ASoC: fix hdmi codec driver contest in S3
by libin.yang＠intel.com 28 Mar '19

28 Mar '19

From: Libin Yang <libin.yang(a)intel.com> In S3, there is a contest between dapm event and getting display power. When resume, the sequence is: hdac_hdmi_runtime_resume() => snd_hdac_display_power() hdac_hdmi_xxx_widget_event() The hdac_hdmi_xxx_widget_event() are based on the power being on. Otherwise, the operation on the codec register will fail. However, in snd_hdac_display_power(), it may sleep because of the mutex_lock() in display driver. In this case, hdac_hdmi_xxx_widget_event() will be called before the power is on. Let's not operate the registers and wait for the power in hdac_hdmi_xxx_widget_event() Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..3c5c122 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -144,6 +144,7 @@ struct hdac_hdmi_priv { int num_pin; int num_cvt; int num_ports; + int display_power; /* 0: power off; 1 power on */ struct mutex pin_mutex; struct hdac_chmap chmap; struct hdac_hdmi_drv_data *drv_data; @@ -742,12 +743,27 @@ static void hdac_hdmi_set_amp(struct hdac_device *hdev, AC_VERB_SET_AMP_GAIN_MUTE, val); } +static int wait_for_display_power(struct hdac_hdmi_priv *hdmi) +{ + int i = 0; + + /* sleep for totally 80ms ~ 200ms should be enough */ + while (i < 40) { + if (!hdmi->display_power) + usleep_range(2000, 5000); + else + return 0; + i++; + } + return -EAGAIN; +} static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", @@ -757,6 +773,11 @@ static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, if (!pcm) return -EIO; + if (wait_for_display_power(hdmi)) { + dev_err(&hdev->dev, "%s: hdmi power is not ready\n", __func__); + return -EAGAIN; + } + /* set the device if pin is mst_capable */ if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; @@ -803,6 +824,11 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, if (!pcm) return -EIO; + if (wait_for_display_power(hdmi)) { + dev_err(&hdev->dev, "%s: hdmi power is not ready\n", __func__); + return -EAGAIN; + } + switch (event) { case SND_SOC_DAPM_PRE_PMU: hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D0); @@ -840,6 +866,7 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, { struct hdac_hdmi_port *port = w->priv; struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int mux_idx; dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", @@ -850,6 +877,11 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, mux_idx = dapm_kcontrol_get_value(kc); + if (wait_for_display_power(hdmi)) { + dev_err(&hdev->dev, "%s: hdmi power is not ready\n", __func__); + return -EAGAIN; + } + /* set the device if pin is mst_capable */ if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; @@ -2068,6 +2100,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_ext_link *hlink = NULL; dev_dbg(dev, "Enter: %s\n", __func__); @@ -2095,6 +2128,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); snd_hdac_display_power(bus, hdev->addr, false); + hdmi->display_power = 0; return 0; } @@ -2102,6 +2136,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) static int hdac_hdmi_runtime_resume(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_link *hlink = NULL; @@ -2128,6 +2163,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + hdmi->display_power = 1; return 0; } #else -- 2.7.4

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[alsa-devel] [PATCH RFC alsa-lib] Drop -I$includedir/alsa from alsa.pc
by Takashi Iwai 28 Mar '19

28 Mar '19

We used to put the additional include path $includedir/alsa in pkgconfig just because some applications have included asoundlib.h like #include <asoundlib.h> although the canonical form should be #include <alsa/asoundlib.h> However, adding this include path is significantly dangerous due to possible conflicts of file names like version.h. It's already the reason to discourage people using alsa.pc for the packages. In this patch, the additional include path from alsa.pc is dropped finally. At the same time, as a rescue plan for the programs including via <asoundlib.h>, a stub header file is provided in include/sound/asoundlib.h. It just includes alsa/asoundlib.h with a warning to suggest for replacing with alsa/asoundlib.h. Actually this is the same file as we install into sys/asoundlib.h, so the whole changes are very minimal here. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/Makefile.am | 1 + utils/alsa.pc.in | 4 +--- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/include/Makefile.am b/include/Makefile.am index 67f32e36c911..fffc8a62e88b 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -96,3 +96,4 @@ AM_CPPFLAGS=-I$(top_srcdir)/include install-data-hook: test -d $(DESTDIR)$(sysincludedir) || mkdir -p $(DESTDIR)$(sysincludedir) $(INSTALL_DATA) $(srcdir)/sys.h $(DESTDIR)$(sysincludedir)/asoundlib.h + $(INSTALL_DATA) $(srcdir)/sys.h $(DESTDIR)/asoundlib.h diff --git a/utils/alsa.pc.in b/utils/alsa.pc.in index 8de9859f42db..444f66d85a34 100644 --- a/utils/alsa.pc.in +++ b/utils/alsa.pc.in @@ -9,6 +9,4 @@ Version: @VERSION@ Requires: Libs: -L${libdir} -lasound Libs.private: @ALSA_DEPLIBS@ -# -I${includedir}/alsa below is just for backward compatibility -# (it was set so mistakely in the older version) -Cflags: -I${includedir} -I${includedir}/alsa +Cflags: -I${includedir} -- 2.16.4

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[alsa-devel] [PATCH] ASoC: mt8183: change supported formats of DL2 and UL1
by Tzung-Bi Shih 28 Mar '19

28 Mar '19

DL2 and UL1 should only provide 16-bit, mono, 8kHz and 16kHz to userspace. Change the formats accordingly. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 43be51bf0329..ad42da1b948c 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -195,9 +195,9 @@ static struct snd_soc_dai_driver mt8183_memif_dai_driver[] = { .playback = { .stream_name = "DL2", .channels_min = 1, - .channels_max = 2, - .rates = MTK_PCM_RATES, - .formats = MTK_PCM_FORMATS, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &mtk_afe_fe_ops, }, @@ -219,9 +219,9 @@ static struct snd_soc_dai_driver mt8183_memif_dai_driver[] = { .capture = { .stream_name = "UL1", .channels_min = 1, - .channels_max = 2, - .rates = MTK_PCM_RATES, - .formats = MTK_PCM_FORMATS, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &mtk_afe_fe_ops, }, -- 2.21.0.392.gf8f6787159e-goog

2 2

[alsa-devel] [PATCH AUTOSEL 3.18 19/41] SoC: imx-sgtl5000: add missing put_device()
by Sasha Levin 27 Mar '19

27 Mar '19

From: Wen Yang <yellowriver2010(a)hotmail.com> [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 1cb22dd034eb..110f3cf361af 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,6 +115,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 45/63] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Sasha Levin 27 Mar '19

27 Mar '19

From: wen yang <yellowriver2010(a)hotmail.com> [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 1b05d1c5d9fd..a32fe14b4687 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -659,6 +659,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 30/63] SoC: imx-sgtl5000: add missing put_device()
by Sasha Levin 27 Mar '19

27 Mar '19

From: Wen Yang <yellowriver2010(a)hotmail.com> [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..8e525f7ac08d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,6 +115,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 60/87] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Sasha Levin 27 Mar '19

27 Mar '19

From: wen yang <yellowriver2010(a)hotmail.com> [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index dffd549a0e2a..705d2524ec31 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 38/87] SoC: imx-sgtl5000: add missing put_device()
by Sasha Levin 27 Mar '19

27 Mar '19

From: Wen Yang <yellowriver2010(a)hotmail.com> [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..8e525f7ac08d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,6 +115,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 084/123] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Sasha Levin 27 Mar '19

27 Mar '19

From: wen yang <yellowriver2010(a)hotmail.com> [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 2db4d0c80d33..393100edd5fd 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 055/123] SoC: imx-sgtl5000: add missing put_device()
by Sasha Levin 27 Mar '19

27 Mar '19

From: Wen Yang <yellowriver2010(a)hotmail.com> [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index b99e0b5e00e9..8e525f7ac08d 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -115,6 +115,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 133/192] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Sasha Levin 27 Mar '19

27 Mar '19

From: wen yang <yellowriver2010(a)hotmail.com> [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 44433b20435c..600d9be9706e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 083/192] SoC: imx-sgtl5000: add missing put_device()
by Sasha Levin 27 Mar '19

27 Mar '19

From: Wen Yang <yellowriver2010(a)hotmail.com> [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..9b9a7ec52905 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -108,6 +108,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.0 183/262] ASoC: fsl-asoc-card: fix object reference leaks in fsl_asoc_card_probe
by Sasha Levin 27 Mar '19

27 Mar '19

From: wen yang <yellowriver2010(a)hotmail.com> [ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Signed-off-by: Wen Yang <yellowriver2010(a)hotmil.com> 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 <festevam(a)gmail.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: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 81f2fe2c6d23..60f87a0d99f4 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -689,6 +689,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); of_node_put(codec_np); + put_device(&cpu_pdev->dev); fail: of_node_put(cpu_np); -- 2.19.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.0 118/262] SoC: imx-sgtl5000: add missing put_device()
by Sasha Levin 27 Mar '19

27 Mar '19

From: Wen Yang <yellowriver2010(a)hotmail.com> [ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ] The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Detected by coccinelle with the following warnings: ./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. ./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device; call of_find_device_by_node on line 105, but without a corresponding object release within this function. Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> 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 <festevam(a)gmail.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: Shawn Guo <shawnguo(a)kernel.org> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Pengutronix Kernel Team <kernel(a)pengutronix.de> Cc: NXP Linux Team <linux-imx(a)nxp.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-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/imx-sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index c29200cf755a..9b9a7ec52905 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -108,6 +108,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto fail; } + put_device(&ssi_pdev->dev); codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); -- 2.19.1

1 0

[alsa-devel] [PATCH -next] ASoC: fix SND_SOC_LOCHNAGAR_SC kconfig warning
by Randy Dunlap 27 Mar '19

27 Mar '19

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kconfig warning for SND_SOC_LOCHNAGAR_SC: WARNING: unmet direct dependencies detected for SND_SOC_LOCHNAGAR_SC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && MFD_LOCHNAGAR [=n] Selected by [m]: - SND_SOC_ALL_CODECS [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && COMPILE_TEST [=y] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Richard Fitzgerald <rf(a)opensource.cirrus.com> Cc: patches(a)opensource.cirrus.com Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20190327.orig/sound/soc/codecs/Kconfig +++ linux-next-20190327/sound/soc/codecs/Kconfig @@ -94,7 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4725B_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C - select SND_SOC_LOCHNAGAR_SC + select SND_SOC_LOCHNAGAR_SC if MFD_LOCHNAGAR select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C

2 1

[alsa-devel] [PATCH 0/4] ALSA: yet more timer cleanup and fixes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, while looking at the old syzkaller reports, I noticed that there are still some possible races around closing the timer object. Here is a patch set to cover it as well as some relevant cleanups / fixes. I haven't set stable to Cc in them since they are really corner cases, and they haven't been tested much. But they should be safe (and easy) to backport, too, if any. Takashi === Takashi Iwai (4): ALSA: timer: Unify timer callback process code ALSA: timer: Make sure to clear pending ack list ALSA: timer: Check ack_list emptiness instead of bit flag ALSA: timer: Make snd_timer_close() really kill pending actions include/sound/timer.h | 1 - sound/core/timer.c | 123 ++++++++++++++++++++++++++++++-------------------- 2 files changed, 74 insertions(+), 50 deletions(-) -- 2.16.4

1 4

[alsa-devel] [PATCH] amixer: define _GNU_SOURCE to get exp10
by Milton Vandersloot 27 Mar '19

27 Mar '19

Hi I get a warning about the implicit declaration of exp10 for amixer (not alsamixer). The same issue was reported at beginning of the year for alsamixer [0,1] and solved by adding -D_GNU_SOURCE to the CFLAGS. (In fact both alsamixer and amixer use the same file alsamixer/volume_mapping.c.) I suggest adding the same CFLAG also to the amixer Makefile.am. Cheers Milton [0] https://patchwork.kernel.org/patch/10765311/ [1] commit 116488e5f2f1b897084bd151381ee254e1cc177d Patch suggestion: >From d31f804d171e7548ecf6e96e96edc7a694a51612 Mon Sep 17 00:00:00 2001 From: Milton Vandersloot <miltonvandersloot(a)fake-box.com> Date: Wed, 27 Mar 2019 17:36:54 +0100 Subject: [PATCH] amixer: define _GNU_SOURCE to get exp10 Get rid of a warning about implicit declarition of exp10, which is behind _GNU_SOURCE as a non-standard feature. The same problem was fixed for alsamixer in commit 116488e5f2f1b897084bd151381ee254e1cc177d. --- amixer/Makefile.am | 1 + 1 file changed, 1 insertion(+) diff --git a/amixer/Makefile.am b/amixer/Makefile.am index 0343c8e..b4526ca 100644 --- a/amixer/Makefile.am +++ b/amixer/Makefile.am @@ -1,3 +1,4 @@ +AM_CFLAGS = -D_GNU_SOURCE AM_CPPFLAGS = -I$(top_srcdir)/include LDADD = -lm # LDFLAGS = -static

1 0

[alsa-devel] [PATCH -next] ASoC: simple-card-utils: remove set but not used variable ''
by YueHaibing 27 Mar '19

27 Mar '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_parse_clk': sound/soc/generic/simple-card-utils.c:164:18: warning: parameter 'dai_name' set but not used [-Wunused-but-set-parameter] It's not used since commit 0580dde59438 ("ASoC: simple-card-utils: add asoc_simple_debug_info()"), so can be removed. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- 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 4ed68348f939..db1458a19985 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -173,10 +173,8 @@ int asoc_simple_parse_clk(struct device *dev, * see * soc-core.c :: snd_soc_init_multicodec() */ - if (dlc) { + if (dlc) dai_of_node = dlc->of_node; - dai_name = dlc->dai_name; - } /* * Parse dai->sysclk come from "clocks = <&xxx>"

3 7

[alsa-devel] [PATCH v3] ASoC: sirf-audio: Remove redundant of_match_node call
by Aditya Pakki 27 Mar '19

27 Mar '19

Unlike other drivers probe method, of_match_node return value is not used or checked. This patch removes the redundant code. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> --- v2: Move the signed by above the version change log v1: A check is unnecessary as match is never used. --- sound/soc/codecs/sirf-audio-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index e424499a8450..e0af21050078 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -461,9 +461,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; struct resource *mem_res; - const struct of_device_id *match; - - match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); -- 2.17.1

3 3

[alsa-devel] [PATCH 0/2] Replace with alloc_pages_exact(), take#2
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a renewed attempt to replace __get_free_pages() usages with alloc_pages_exact() in sound tree. At this time, the incorrectly leftover of __GFP_COMP flag is removed altogether. thanks, Takashi === Takashi Iwai (2): ALSA: Replace snd_malloc_pages() and snd_free_pages() with standard helpers, take#2 ALSA: us122l: Use alloc_pages_exact() include/sound/memalloc.h | 4 ---- sound/core/memalloc.c | 53 +++-------------------------------------- sound/core/pcm.c | 14 +++++------ sound/usb/usx2y/usX2Yhwdep.c | 3 ++- sound/usb/usx2y/usb_stream.c | 20 +++++++--------- sound/usb/usx2y/usbusx2y.c | 3 ++- sound/usb/usx2y/usx2yhwdeppcm.c | 6 +++-- 7 files changed, 26 insertions(+), 77 deletions(-) -- 2.16.4

2 6

[alsa-devel] [PATCH v12 0/5] Media Device Allocator API
by Shuah Khan 27 Mar '19

27 Mar '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - This patch series in tested on 5.1-rc2 and addresses the compile errors, warns, and sparse errors reported by Hans Verkuil and Dan Carpenter. Changes since v11: - Patch 1: Add CONFIG_MODULES dependency in media_dev_allocator files. to fix compile errors when CONFIG_MODULES is disabled. - Patch 2, 3: No changes. - Patch 4: Fix sparse error reported by Dan Carpenter. - Patch 5: Fix warns found by Hans Verkuil. - v11 was tested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. The test plan used for testing resource sharing could serve as a regression test plan and the test results can be found at: https://docs.google.com/document/d/1XMs3HYgLiHby6xVIvIxv75KSXAAN3F4uUpw2uLu… - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. - v9 was tested on 4.20-rc6. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (5): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 142 +++++++++++ drivers/media/usb/au0828/au0828-core.c | 194 +++++++++++---- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 54 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 327 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 872 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

3 9

[alsa-devel] [RFC 0/2] Add MICFIL DAI support
by Cosmin Samoila 27 Mar '19

27 Mar '19

PDM is a popular way to deliver audio from microphones to the processor in several applications, such as mobile telephones. However, current digital-audio systems use multibit audio signal (also known as multibit PCM) to represent the signal. For this purpose a set of FIR, CIC or/and Half Band filters are usually implemented on DSPs or software. This block implements the required digital interface to provide a 16-bit audio signal from a PDM microphone bitstream in a configurable output sampling rate. The implementation of this digital interface is based on the application of digital signal processing techniques in hardware. The PDM Microphone Interface architecture was designed to gate saving and minimal power consumption. It implements a bunch of filters to transform a 1-bit PDM bitstream to a 16-bit PCM signal in the audio band. To avoid aliasing frequencies in passband, the overall filter has 80 dB stopband attenuation and passband ripple less than 0.2dB. The whole module is implemented to work in a multichannel mode. All channels have the same configuration but each input channel could be turned on/off independently. Cosmin-Gabriel Samoila (2): ASoC: micfil: Add bindings for MICFIL DAI ASoC: Add MICFIL SoC Digital Audio Interface driver. .../devicetree/bindings/sound/fsl,micfil.txt | 38 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_micfil.c | 2451 ++++++++++++++++++++ sound/soc/fsl/fsl_micfil.h | 317 +++ 5 files changed, 2817 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 sound/soc/fsl/fsl_micfil.c create mode 100644 sound/soc/fsl/fsl_micfil.h -- 2.7.4

6 18

[alsa-devel] [PATCH alsa-plugins] configure: Fix unexpanded ALSA_DATA_DIR
by Takashi Iwai 27 Mar '19

27 Mar '19

When no --with-alsadatadir is given, configure scripts takes $datadir/alsa as the default location, while it's expanded to $prefix/data/alsa. Although ALSA_DATA_DIR is set via AC_DEFINE_UNQUOTED(), this expands only $alsadatadir itself and not about the $prefix in the content. For resolving this, we need to do eval twice. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 1be4a70c803c..5b131e704c2f 100644 --- a/configure.ac +++ b/configure.ac @@ -211,7 +211,7 @@ if test -z "$alsadatadir"; then eval dir="$datadir" case "$dir" in /*) ;; - *) dir="$dir" + *) eval dir="$dir" esac alsadatadir="$dir/alsa" fi -- 2.16.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_audmix: Fix kbuild failure
by Viorel Suman 27 Mar '19

27 Mar '19

The format in dev_dbg function must be a constant. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_audmix.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 3356cb6..dabde03 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -90,7 +90,7 @@ static int fsl_audmix_state_trans(struct snd_soc_component *comp, struct fsl_audmix *priv = snd_soc_component_get_drvdata(comp); /* Enforce all required TDMs are started */ if ((priv->tdms & prm.tdms) != prm.tdms) { - dev_dbg(comp->dev, prm.msg); + dev_dbg(comp->dev, "%s", prm.msg); return -EINVAL; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: sprd: Add Spreadtrum audio compress offload support
by Baolin Wang 27 Mar '19

27 Mar '19

We use 2-stage DMA mode to support Spreadtrum audio compress offload, which means we use one DMA source channel to transfer data from IRAM buffer to the DSP fifo to do decoding/encoding, once IRAM buffer is empty by transferring done, another DMA destination channel will be triggered automatically to start to transfer data from DDR buffer to the IRAM buffer. This can reduce the AP subsystem wakeup times to save power. Co-developed-by: Yintang Ren <yintang.ren(a)unisoc.com> Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> --- sound/soc/sprd/Kconfig | 1 + sound/soc/sprd/Makefile | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 675 ++++++++++++++++++++++++++++++++++++ sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/sprd/sprd-pcm-dma.h | 43 +++ 5 files changed, 721 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sprd/sprd-pcm-compress.c diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 43ece7d..3b1eb32 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -1,6 +1,7 @@ config SND_SOC_SPRD tristate "SoC Audio for the Spreadtrum SoC chips" depends on ARCH_SPRD || COMPILE_TEST + select SND_SOC_COMPRESS help Say Y or M if you want to add support for codecs attached to the Spreadtrum SoCs' Audio interfaces. diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile index 47620e5..fa1d547 100644 --- a/sound/soc/sprd/Makefile +++ b/sound/soc/sprd/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0 # Spreadtrum Audio Support -obj-$(CONFIG_SND_SOC_SPRD) += sprd-pcm-dma.o +obj-$(CONFIG_SND_SOC_SPRD) += sprd-pcm-dma.o sprd-pcm-compress.o diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c new file mode 100644 index 0000000..dc84257 --- /dev/null +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -0,0 +1,675 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (C) 2019 Spreadtrum Communications Inc. + +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/dma/sprd-dma.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/compress_driver.h> + +#include "sprd-pcm-dma.h" + +#define SPRD_COMPR_DMA_CHANS 2 + +/* Default values if userspace does not set */ +#define SPRD_COMPR_MIN_FRAGMENT_SIZE SZ_8K +#define SPRD_COMPR_MAX_FRAGMENT_SIZE SZ_128K +#define SPRD_COMPR_MIN_NUM_FRAGMENTS 4 +#define SPRD_COMPR_MAX_NUM_FRAGMENTS 64 + +/* DSP FIFO size */ +#define SPRD_COMPR_MCDT_EMPTY_WMK 0 +#define SPRD_COMPR_MCDT_FIFO_SIZE 512 + +/* Stage 0 IRAM buffer size definition */ +#define SPRD_COMPR_IRAM_BUF_SIZE SZ_32K +#define SPRD_COMPR_IRAM_INFO_SIZE (sizeof(struct sprd_compr_playinfo)) +#define SPRD_COMPR_IRAM_LINKLIST_SIZE (1024 - SPRD_COMPR_IRAM_INFO_SIZE) +#define SPRD_COMPR_IRAM_SIZE (SPRD_COMPR_IRAM_BUF_SIZE + \ + SPRD_COMPR_IRAM_INFO_SIZE + \ + SPRD_COMPR_IRAM_LINKLIST_SIZE) + +/* Stage 1 DDR buffer size definition */ +#define SPRD_COMPR_AREA_BUF_SIZE SZ_2M +#define SPRD_COMPR_AREA_LINKLIST_SIZE 1024 +#define SPRD_COMPR_AREA_SIZE (SPRD_COMPR_AREA_BUF_SIZE + \ + SPRD_COMPR_AREA_LINKLIST_SIZE) + +struct sprd_compr_dma { + struct dma_chan *chan; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + dma_addr_t phys; + void *virt; + int trans_len; +}; + +/* + * The Spreadtrum Audio compress offload mode will use 2-stage DMA transfer to + * save power. That means we can request 2 dma channels, one for source channel, + * and another one for destination channel. Once the source channel's transaction + * is done, it will trigger the destination channel's transaction automatically + * by hardware signal. + * + * For 2-stage DMA transfer, we can allocate 2 buffers: IRAM buffer (always + * power-on) and DDR buffer. The source channel will transfer data from IRAM + * buffer to the DSP fifo to decoding/encoding, once IRAM buffer is empty by + * transferring done, the destination channel will start to transfer data from + * DDR buffer to IRAM buffer. + * + * Since the DSP fifo is only 512B, IRAM buffer is allocated by 32K, and DDR + * buffer is larger to 2M. That means only the IRAM 32k data is transferred + * done, we can wake up the AP system to transfer data from DDR to IRAM, and + * other time the AP system can be suspended to save power. + */ +struct sprd_compr_stream { + struct snd_compr_stream *cstream; + struct sprd_compr_ops *compr_ops; + struct sprd_compr_dma dma[SPRD_COMPR_DMA_CHANS]; + + /* DMA engine channel number */ + int num_channels; + + /* Stage 0 IRAM buffer */ + struct snd_dma_buffer iram_buffer; + /* Stage 1 DDR buffer */ + struct snd_dma_buffer compr_buffer; + + /* DSP play information IRAM buffer */ + dma_addr_t info_phys; + void *info_area; + int info_size; + + /* Data size copied to IRAM buffer */ + int copied_total; + /* Total received data size from userspace */ + int received_total; + /* Stage 0 IRAM buffer received data size */ + int received_stage0; + /* Stage 1 DDR buffer received data size */ + int received_stage1; + /* Stage 1 DDR buffer pointer */ + int stage1_pointer; +}; + +static int sprd_platform_compr_trigger(struct snd_compr_stream *cstream, + int cmd); + +static void sprd_platform_compr_drain_notify(void *arg) +{ + struct snd_compr_stream *cstream = arg; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + + memset(stream->info_area, 0, sizeof(struct sprd_compr_playinfo)); + + if (cstream) + snd_compr_drain_notify(cstream); +} + +static void sprd_platform_compr_dma_complete(void *data) +{ + struct snd_compr_stream *cstream = data; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct sprd_compr_dma *dma = &stream->dma[1]; + + /* Update data size copied to IRAM buffer */ + stream->copied_total += dma->trans_len; + if (stream->copied_total > stream->received_total) + stream->copied_total = stream->received_total; + + snd_compr_fragment_elapsed(cstream); +} + +static int sprd_platform_compr_dma_config(struct snd_compr_stream *cstream, + struct snd_compr_params *params, + int channel) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_pcm_dma_params *dma_params = data->dma_params; + struct sprd_compr_dma *dma = &stream->dma[channel]; + struct dma_slave_config config = { }; + struct sprd_dma_linklist link = { }; + enum dma_transfer_direction dir; + struct scatterlist *sg, *sgt; + enum dma_slave_buswidth bus_width; + int period, period_cnt, sg_num = 2; + dma_addr_t src_addr, dst_addr; + unsigned long flags; + int ret, j; + + if (!dma_params) { + dev_err(dev, "no dma parameters setting\n"); + return -EINVAL; + } + + dma->chan = dma_request_slave_channel(dev, + dma_params->chan_name[channel]); + if (!dma->chan) { + dev_err(dev, "failed to request dma channel\n"); + return -ENODEV; + } + + sgt = sg = devm_kcalloc(dev, sg_num, sizeof(*sg), GFP_KERNEL); + if (!sg) { + ret = -ENOMEM; + goto sg_err; + } + + switch (channel) { + case 0: + bus_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + period = (SPRD_COMPR_MCDT_FIFO_SIZE - SPRD_COMPR_MCDT_EMPTY_WMK) * 4; + period_cnt = params->buffer.fragment_size / period; + src_addr = stream->iram_buffer.addr; + dst_addr = dma_params->dev_phys[channel]; + flags = SPRD_DMA_FLAGS(SPRD_DMA_SRC_CHN1, + SPRD_DMA_TRANS_DONE_TRG, + SPRD_DMA_FRAG_REQ, + SPRD_DMA_TRANS_INT); + break; + + case 1: + bus_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + period = params->buffer.fragment_size; + period_cnt = params->buffer.fragments; + src_addr = stream->compr_buffer.addr; + dst_addr = stream->iram_buffer.addr; + flags = SPRD_DMA_FLAGS(SPRD_DMA_DST_CHN1, + SPRD_DMA_TRANS_DONE_TRG, + SPRD_DMA_FRAG_REQ, + SPRD_DMA_TRANS_INT); + break; + + default: + ret = -EINVAL; + goto config_err; + } + + dma->trans_len = period * period_cnt; + + config.src_maxburst = period; + config.src_addr_width = bus_width; + config.dst_addr_width = bus_width; + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + config.src_addr = src_addr; + config.dst_addr = dst_addr; + dir = DMA_MEM_TO_DEV; + } else { + config.src_addr = dst_addr; + config.dst_addr = src_addr; + dir = DMA_DEV_TO_MEM; + } + + sg_init_table(sgt, sg_num); + for (j = 0; j < sg_num; j++, sgt++) { + sg_dma_len(sgt) = dma->trans_len; + sg_dma_address(sgt) = dst_addr; + } + + /* + * Configure the link-list address for the DMA engine link-list + * mode. + */ + link.virt_addr = (unsigned long)dma->virt; + link.phy_addr = dma->phys; + + ret = dmaengine_slave_config(dma->chan, &config); + if (ret) { + dev_err(dev, + "failed to set slave configuration: %d\n", ret); + goto config_err; + } + + /* + * We configure the DMA request mode, interrupt mode, channel + * mode and channel trigger mode by the flags. + */ + dma->desc = dma->chan->device->device_prep_slave_sg(dma->chan, sg, + sg_num, dir, + flags, &link); + if (!dma->desc) { + dev_err(dev, "failed to prepare slave sg\n"); + ret = -ENOMEM; + goto config_err; + } + + /* Only channel 1 transfer can wake up the AP system. */ + if (!params->no_wake_mode && channel == 1) { + dma->desc->callback = sprd_platform_compr_dma_complete; + dma->desc->callback_param = cstream; + } + + devm_kfree(dev, sg); + + return 0; + +config_err: + devm_kfree(dev, sg); +sg_err: + dma_release_channel(dma->chan); + return ret; +} + +static int sprd_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + struct sprd_compr_params compr_params = { }; + int ret; + + /* + * Configure the DMA engine 2-stage transfer mode. Channel 1 set as the + * destination channel, and channel 0 set as the source channel, that + * means once the source channel's transaction is done, it will trigger + * the destination channel's transaction automatically. + */ + ret = sprd_platform_compr_dma_config(cstream, params, 1); + if (ret) { + dev_err(dev, "faied to config stage 1 DMA: %d\n", ret); + return ret; + } + + ret = sprd_platform_compr_dma_config(cstream, params, 0); + if (ret) { + dev_err(dev, "faied to config stage 0 DMA: %d\n", ret); + goto config_err; + } + + compr_params.direction = cstream->direction; + compr_params.sample_rate = params->codec.sample_rate; + compr_params.channels = stream->num_channels; + compr_params.info_phys = stream->info_phys; + compr_params.info_size = stream->info_size; + compr_params.rate = params->codec.bit_rate; + compr_params.format = params->codec.id; + + ret = stream->compr_ops->set_params(cstream->direction, &compr_params); + if (ret) { + dev_err(dev, "failed to set parameters: %d\n", ret); + goto params_err; + } + + return 0; + +params_err: + dma_release_channel(stream->dma[0].chan); +config_err: + dma_release_channel(stream->dma[1].chan); + return ret; +} + +static int sprd_platform_compr_open(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + struct sprd_compr_data *data = snd_soc_dai_get_drvdata(rtd->cpu_dai); + struct sprd_compr_stream *stream; + struct sprd_compr_callback cb; + int stream_id = cstream->direction, ret; + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + stream = devm_kzalloc(dev, sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->cstream = cstream; + stream->num_channels = 2; + stream->compr_ops = data->ops; + + /* + * Allocate the stage 0 IRAM buffer size, including the DMA 0 + * link-list size and play information of DSP address size. + */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_IRAM, dev, + SPRD_COMPR_IRAM_SIZE, &stream->iram_buffer); + if (ret < 0) + goto err_iram; + + /* Use to save link-list configuration for DMA 0. */ + stream->dma[0].virt = stream->iram_buffer.area + SPRD_COMPR_IRAM_SIZE; + stream->dma[0].phys = stream->iram_buffer.addr + SPRD_COMPR_IRAM_SIZE; + + /* Use to update the current data offset of DSP. */ + stream->info_phys = stream->iram_buffer.addr + SPRD_COMPR_IRAM_SIZE + + SPRD_COMPR_IRAM_LINKLIST_SIZE; + stream->info_area = stream->iram_buffer.area + SPRD_COMPR_IRAM_SIZE + + SPRD_COMPR_IRAM_LINKLIST_SIZE; + stream->info_size = SPRD_COMPR_IRAM_INFO_SIZE; + + /* + * Allocate the stage 1 DDR buffer size, including the DMA 1 link-list + * size. + */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, + SPRD_COMPR_AREA_SIZE, &stream->compr_buffer); + if (ret < 0) + goto err_compr; + + /* Use to save link-list configuration for DMA 1. */ + stream->dma[1].virt = stream->compr_buffer.area + SPRD_COMPR_AREA_SIZE; + stream->dma[1].phys = stream->compr_buffer.addr + SPRD_COMPR_AREA_SIZE; + + cb.drain_notify = sprd_platform_compr_drain_notify; + cb.drain_data = cstream; + ret = stream->compr_ops->open(stream_id, &cb); + if (ret) { + dev_err(dev, "failed to open compress platform: %d\n", ret); + goto err_open; + } + + runtime->private_data = stream; + return 0; + +err_open: + snd_dma_free_pages(&stream->compr_buffer); +err_compr: + snd_dma_free_pages(&stream->iram_buffer); +err_iram: + devm_kfree(dev, stream); + + return ret; +} + +static int sprd_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + int stream_id = cstream->direction, i; + + for (i = 0; i < stream->num_channels; i++) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) { + dma_release_channel(dma->chan); + dma->chan = NULL; + } + } + + snd_dma_free_pages(&stream->compr_buffer); + snd_dma_free_pages(&stream->iram_buffer); + + stream->compr_ops->close(stream_id); + + devm_kfree(dev, stream); + return 0; +} + +static int sprd_platform_compr_trigger(struct snd_compr_stream *cstream, + int cmd) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct device *dev = component->dev; + int channels = stream->num_channels, ret = 0, i; + int stream_id = cstream->direction; + + if (cstream->direction != SND_COMPRESS_PLAYBACK) { + dev_err(dev, "unsupported compress direction\n"); + return -EINVAL; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (!dma->desc) + continue; + + dma->cookie = dmaengine_submit(dma->desc); + ret = dma_submit_error(dma->cookie); + if (ret) { + dev_err(dev, "failed to submit request: %d\n", + ret); + return ret; + } + } + + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dma_async_issue_pending(dma->chan); + } + + ret = stream->compr_ops->start(stream_id); + break; + + case SNDRV_PCM_TRIGGER_STOP: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dmaengine_terminate_async(dma->chan); + } + + stream->copied_total = 0; + stream->stage1_pointer = 0; + stream->received_total = 0; + stream->received_stage0 = 0; + stream->received_stage1 = 0; + + ret = stream->compr_ops->stop(stream_id); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dmaengine_pause(dma->chan); + } + + ret = stream->compr_ops->pause(stream_id); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + for (i = channels - 1; i >= 0; i--) { + struct sprd_compr_dma *dma = &stream->dma[i]; + + if (dma->chan) + dmaengine_resume(dma->chan); + } + + ret = stream->compr_ops->pause_release(stream_id); + break; + + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + case SND_COMPR_TRIGGER_DRAIN: + ret = stream->compr_ops->drain(stream->received_total); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int sprd_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + struct sprd_compr_playinfo *info = + (struct sprd_compr_playinfo *)stream->info_area; + + tstamp->copied_total = stream->copied_total; + tstamp->pcm_io_frames = info->current_data_offset; + + return 0; +} + +static int sprd_platform_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct sprd_compr_stream *stream = runtime->private_data; + int avail_bytes, data_count = count; + void *dst; + + /* + * We usually set fragment size as 32K, and the stage 0 IRAM buffer + * size is 32K too. So if now the received data size of the stage 0 + * IRAM buffer is less than 32K, that means we have some available + * spaces for the stage 0 IRAM buffer. + */ + if (stream->received_stage0 < runtime->fragment_size) { + avail_bytes = runtime->fragment_size - stream->received_stage0; + dst = stream->iram_buffer.area + stream->received_stage0; + + if (avail_bytes >= data_count) { + /* + * Copy data to the stage 0 IRAM buffer directly if + * spaces are enough. + */ + if (copy_from_user(dst, buf, data_count)) + return -EFAULT; + + stream->received_stage0 += data_count; + stream->copied_total += data_count; + goto copy_done; + } else { + /* + * If the data count is larger than the available spaces + * of the the stage 0 IRAM buffer, we should copy one + * partial data to the stage 0 IRAM buffer, and copy + * the left to the stage 1 DDR buffer. + */ + if (copy_from_user(dst, buf, avail_bytes)) + return -EFAULT; + + data_count -= avail_bytes; + stream->received_stage0 += avail_bytes; + stream->copied_total += avail_bytes; + buf += avail_bytes; + } + } + + /* + * Copy data to the stage 1 DDR buffer if no spaces for the stage 0 IRAM + * buffer. + */ + dst = stream->compr_buffer.area + stream->stage1_pointer; + if (data_count < stream->compr_buffer.bytes - stream->stage1_pointer) { + if (copy_from_user(dst, buf, data_count)) + return -EFAULT; + + stream->stage1_pointer += data_count; + } else { + avail_bytes = stream->compr_buffer.bytes - stream->stage1_pointer; + + if (copy_from_user(dst, buf, avail_bytes)) + return -EFAULT; + + if (copy_from_user(stream->compr_buffer.area, buf + avail_bytes, + data_count - avail_bytes)) + return -EFAULT; + + stream->stage1_pointer = data_count - avail_bytes; + } + + stream->received_stage1 += data_count; + +copy_done: + /* Update the copied data size. */ + stream->received_total += count; + return count; +} + +static int sprd_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + caps->direction = cstream->direction; + caps->min_fragment_size = SPRD_COMPR_MIN_FRAGMENT_SIZE; + caps->max_fragment_size = SPRD_COMPR_MAX_FRAGMENT_SIZE; + caps->min_fragments = SPRD_COMPR_MIN_NUM_FRAGMENTS; + caps->max_fragments = SPRD_COMPR_MAX_NUM_FRAGMENTS; + caps->num_codecs = 2; + caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_AAC; + + return 0; +} + +static int +sprd_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + switch (codec->codec) { + case SND_AUDIOCODEC_MP3: + codec->num_descriptors = 2; + codec->descriptor[0].max_ch = 2; + codec->descriptor[0].bit_rate[0] = 320; + codec->descriptor[0].bit_rate[1] = 128; + codec->descriptor[0].num_bitrates = 2; + codec->descriptor[0].profiles = 0; + codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO; + codec->descriptor[0].formats = 0; + break; + + case SND_AUDIOCODEC_AAC: + codec->num_descriptors = 2; + codec->descriptor[1].max_ch = 2; + codec->descriptor[1].bit_rate[0] = 320; + codec->descriptor[1].bit_rate[1] = 128; + codec->descriptor[1].num_bitrates = 2; + codec->descriptor[1].profiles = 0; + codec->descriptor[1].modes = 0; + codec->descriptor[1].formats = 0; + break; + + default: + return -EINVAL; + } + + return 0; +} + +const struct snd_compr_ops sprd_platform_compr_ops = { + .open = sprd_platform_compr_open, + .free = sprd_platform_compr_free, + .set_params = sprd_platform_compr_set_params, + .trigger = sprd_platform_compr_trigger, + .pointer = sprd_platform_compr_pointer, + .copy = sprd_platform_compr_copy, + .get_caps = sprd_platform_compr_get_caps, + .get_codec_caps = sprd_platform_compr_get_codec_caps, +}; + +MODULE_DESCRIPTION("Spreadtrum ASoC Compress Platform Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:compress-platform"); diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index cbb27c4..9be6d4b 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -13,7 +13,6 @@ #include "sprd-pcm-dma.h" -#define DRV_NAME "sprd_pcm_dma" #define SPRD_PCM_DMA_LINKLIST_SIZE 64 #define SPRD_PCM_DMA_BRUST_LEN 640 @@ -524,6 +523,7 @@ static void sprd_pcm_free(struct snd_pcm *pcm) static const struct snd_soc_component_driver sprd_soc_component = { .name = DRV_NAME, .ops = &sprd_pcm_ops, + .compr_ops = &sprd_platform_compr_ops, .pcm_new = sprd_pcm_new, .pcm_free = sprd_pcm_free, }; diff --git a/sound/soc/sprd/sprd-pcm-dma.h b/sound/soc/sprd/sprd-pcm-dma.h index d85a34f..08e9fdba 100644 --- a/sound/soc/sprd/sprd-pcm-dma.h +++ b/sound/soc/sprd/sprd-pcm-dma.h @@ -3,8 +3,11 @@ #ifndef __SPRD_PCM_DMA_H #define __SPRD_PCM_DMA_H +#define DRV_NAME "sprd_pcm_dma" #define SPRD_PCM_CHANNEL_MAX 2 +extern const struct snd_compr_ops sprd_platform_compr_ops; + struct sprd_pcm_dma_params { dma_addr_t dev_phys[SPRD_PCM_CHANNEL_MAX]; u32 datawidth[SPRD_PCM_CHANNEL_MAX]; @@ -12,4 +15,44 @@ struct sprd_pcm_dma_params { const char *chan_name[SPRD_PCM_CHANNEL_MAX]; }; +struct sprd_compr_playinfo { + int total_time; + int current_time; + int total_data_length; + int current_data_offset; +}; + +struct sprd_compr_params { + u32 direction; + u32 rate; + u32 sample_rate; + u32 channels; + u32 format; + u32 period; + u32 periods; + u32 info_phys; + u32 info_size; +}; + +struct sprd_compr_callback { + void (*drain_notify)(void *data); + void *drain_data; +}; + +struct sprd_compr_ops { + int (*open)(int str_id, struct sprd_compr_callback *cb); + int (*close)(int str_id); + int (*start)(int str_id); + int (*stop)(int str_id); + int (*pause)(int str_id); + int (*pause_release)(int str_id); + int (*drain)(int received_total); + int (*set_params)(int str_id, struct sprd_compr_params *params); +}; + +struct sprd_compr_data { + struct sprd_compr_ops *ops; + struct sprd_pcm_dma_params *dma_params; +}; + #endif /* __SPRD_PCM_DMA_H */ -- 1.7.9.5

1 0

[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

3 40

[alsa-devel] [PATCH] ASoC: core: Fix use-after-free after deferred card registration
by Guenter Roeck 26 Mar '19

26 Mar '19

If snd_soc_register_card() fails because one of its links fails to instantiate with -EPROBE_DEFER, and the to-be-registered link is a legacy link, a subsequent retry will trigger a use-after-free and quite often a system crash. Example: byt-max98090 byt-max98090: ASoC: failed to init link Baytrail Audio byt-max98090 byt-max98090: snd_soc_register_card failed -517 .... BUG: KASAN: use-after-free in snd_soc_init_platform+0x233/0x312 Read of size 8 at addr ffff888067c43070 by task kworker/1:1/23 snd_soc_init_platform() allocates memory attached to the card device. This memory is released when the card device is released. However, the pointer to the memory (dai_link->platforms) is only cleared from soc_cleanup_platform(), which is called from soc_cleanup_card_resources(), but not if snd_soc_register_card() fails early. Add the missing call to soc_cleanup_platform() in the error handling code of snd_soc_register_card() to fix the problem. Fixes: 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error") Cc: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..6bf9884d0863 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2799,6 +2799,7 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + soc_cleanup_platform(card); mutex_unlock(&client_mutex); return ret; } -- 2.7.4

2 2

[alsa-devel] [RESEND] [-, JACK, plugin, 1/1] configure: use /etc instead of /etc
by Alex Ivanov 26 Mar '19

26 Mar '19

Could you please test this change again? I need it for my maintainer work. Thank you! Use $sysconfdir as prefix for ALSA_LCONF_DIR by default. Otherwise install fails on non-FHS distros. Signed-off-by: Alex Ivanov <gnidorah(a)ya.ru> --- configure.ac | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 5b80585..1be4a70 100644 --- a/configure.ac +++ b/configure.ac @@ -237,7 +237,12 @@ AC_ARG_WITH(alsalconfdir, [path where ALSA local add-on config files are stored]), alsalconfdir="$withval", alsalconfdir="") if test -z "$alsalconfdir"; then - alsalconfdir="/etc/alsa/conf.d" + eval dir="$sysconfdir" + case "$dir" in + /*) ;; + *) dir="$dir" + esac + alsalconfdir="$dir/alsa/conf.d" fi AC_DEFINE_UNQUOTED(ALSA_LCONF_DIR, "$alsalconfdir", [directory containing local ALSA add-on config files]) ALSA_LCONF_DIR="$alsalconfdir" -- 2.19.2

1 0

[alsa-devel] [asoc:for-5.2 84/86] include/linux/dynamic_debug.h:80:13: error: initializer element is not constant
by kbuild test robot 26 Mar '19

26 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: b86ef5367761603df5f66ce08fb014b991f6b51d commit: be1df61cf06efb355c90702e46b8d46f055acb4e [84/86] ASoC: fsl: Add Audio Mixer CPU DAI driver config: sparc64-allmodconfig (attached as .config) compiler: sparc64-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 be1df61cf06efb355c90702e46b8d46f055acb4e # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sparc64 All errors (new ones prefixed by >>): In file included from include/linux/printk.h:330:0, from include/linux/kernel.h:15, from include/linux/clk.h:16, from sound/soc//fsl/fsl_audmix.c:8: sound/soc//fsl/fsl_audmix.c: In function 'fsl_audmix_state_trans': >> include/linux/dynamic_debug.h:80:13: error: initializer element is not constant .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc//fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ include/linux/dynamic_debug.h:80:13: note: (near initialization for '__UNIQUE_ID_ddebug14.format') .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc//fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ -- In file included from include/linux/printk.h:330:0, from include/linux/kernel.h:15, from include/linux/clk.h:16, from sound/soc/fsl/fsl_audmix.c:8: sound/soc/fsl/fsl_audmix.c: In function 'fsl_audmix_state_trans': >> include/linux/dynamic_debug.h:80:13: error: initializer element is not constant .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc/fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ include/linux/dynamic_debug.h:80:13: note: (near initialization for '__UNIQUE_ID_ddebug14.format') .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc/fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ vim +80 include/linux/dynamic_debug.h ffa10cb4 Jason Baron 2011-08-11 68 b9075fa9 Joe Perches 2011-10-31 69 extern __printf(3, 4) 906d2015 Joe Perches 2014-09-24 70 void __dynamic_netdev_dbg(struct _ddebug *descriptor, ffa10cb4 Jason Baron 2011-08-11 71 const struct net_device *dev, b9075fa9 Joe Perches 2011-10-31 72 const char *fmt, ...); ffa10cb4 Jason Baron 2011-08-11 73 2bdde670 Rasmus Villemoes 2019-03-07 74 #define DEFINE_DYNAMIC_DEBUG_METADATA(name, fmt) \ c0d2af63 Joe Perches 2012-10-18 75 static struct _ddebug __aligned(8) \ 07613b0b Jason Baron 2011-10-04 76 __attribute__((section("__verbose"))) name = { \ 07613b0b Jason Baron 2011-10-04 77 .modname = KBUILD_MODNAME, \ 07613b0b Jason Baron 2011-10-04 78 .function = __func__, \ 07613b0b Jason Baron 2011-10-04 79 .filename = __FILE__, \ 07613b0b Jason Baron 2011-10-04 @80 .format = (fmt), \ 07613b0b Jason Baron 2011-10-04 81 .lineno = __LINE__, \ 07613b0b Jason Baron 2011-10-04 82 .flags = _DPRINTK_FLAGS_DEFAULT, \ 2bdde670 Rasmus Villemoes 2019-03-07 83 _DPRINTK_KEY_INIT \ 07613b0b Jason Baron 2011-10-04 84 } 07613b0b Jason Baron 2011-10-04 85 :::::: The code at line 80 was first introduced by commit :::::: 07613b0b5ef8570033aa806d1731dce599862223 dynamic_debug: consolidate repetitive struct _ddebug descriptor definitions :::::: TO: Jason Baron <jbaron(a)redhat.com> :::::: CC: Greg Kroah-Hartman <gregkh(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] snd_pcm_writei 0 frames
by Giuliano Zannetti - ART S.p.A. 26 Mar '19

26 Mar '19

Hi, I've the following configuration in my asound.conf: |-----| - - > |-----| - - > |-----| - - > |-----| - - > |-----| - - > |----------| |plug | | | - - > | | - - > | | - - > | | - - > | (output) | |-----| - - > | | - - > | | - - > | | - - > | | - - > | hw | | | - - > | | - - > | | - - > | | - - > | | | | | | |route| - - > |dmix | - - > | | | | | | | | - - > | | - - > | | | | | | | | - - > | | - - > | | | | | | |-----| - - > |-----| - - > |----------| |route| |multi| | | - - > | | | | - - > | | - - > |----| - - > |----| - - > |----------| | | - - > | | - - > | | - - > | | - - > |(loopback)| |-----| - - > | | - - > |plug| - - > |dmix| - - > | hw | |-----| - - > |----| - - > |----| - - > |----------| As you can see I'm using a loopback interface. I open the pcm plug in the block mode to write. writtenFrames = snd_pcm_writei(handle, bufferToWrite, bufferToWriteSize); Using the configuration above I have several "writes" with 0 written frames. It seems like the snd_pcm_writei does not hold until data will be written. In general, 0 written frames sounds weird to me, because snd_pcm_writei should block itself if no space is available in the circular buffer right? So, my questions are two: 1. Writing 0 frames is an expected behaviour (regardless of my configuration in the asound.conf)? In what scenarios can I have that? 2. How my configuration could trigger this behaviour? Best Regards Giuliano Zannetti

2 2

[alsa-devel] [PATCH] ALSA: emux: Add support of loading GUS-patch
by Takashi Iwai 26 Mar '19

26 Mar '19

It's a feature request for the ancient sutff, but it's still valid; the loading of a GUS-patch isn't available via hwdep device although it's supported over OSS sequencer. The only missing piece is the call of snd_soundfont_load_guspatch() in synth emux hwdep code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/synth/emux/emux_hwdep.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index d9fcae071b47..fae48d108b97 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -39,6 +39,11 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg) if (copy_from_user(&patch, arg, sizeof(patch))) return -EFAULT; + if (patch.key == GUS_PATCH) + return snd_soundfont_load_guspatch(emu->sflist, arg, + patch.len + sizeof(patch), + TMP_CLIENT_ID); + if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) { err = snd_soundfont_load(emu->sflist, arg, patch.len + sizeof(patch), TMP_CLIENT_ID); -- 2.16.4

1 0

[alsa-devel] Common Mode Voltage Setting
by Annaliese McDermond 26 Mar '19

26 Mar '19

I have an architectural question regarding some work I’m doing. We have a product that hooks the Raspberry Pi to various amateur radio rigs via a sound card interface. The input characteristics of those radios vary widely in the amount of signal they need to drive the proper RF output and such. One of the setting s I’ve implemented to try to help our customers control this is changing the Common Mode voltage on the tlv320. There are a couple of options, but the gist is that one will allow higher output levels without clipping. We need to keep the flexibility of having the lower setting because there are some radios that require as little drive as possible. My question is really surrounding where the appropriate setting for this would be. I could put it in the device tree, but that requires adding entries, pushing this past the DT folks and it becomes something that a user has to reboot to change settings on. I think it probably more appropriately goes as a mixer control so that it can be changed on the fly without reboot and is a dynamic parameter. I just want to make sure there’s no strong preference here on the “right way” it should be done or if I’m missing a way to set this that’s a “none of the above.” Thanks! -- Annaliese McDermond nh6z(a)nh6z.net

1 0

[alsa-devel] NHLT Specification
by Iker Perez del Palomar 26 Mar '19

26 Mar '19

Hello, My name is Iker Perez, I am contacting you because at the moment I am working with NHLT at my company, Codethink. This is the first time we have been involved with it, so we have been trying to find a specification or a document that could help us understand it better. We tried the Intel website [1] but it hasn't got the document it states is linked to it. I was wondering if someone could point us toward any documentation you are aware of? Thank you, Regards, Iker -- Iker Perez del Palomar, Software Engineer Codethink Ltd 35 Dale St, Manchester M1 2HF http://www.codethink.co.uk/ Manchester, M1 2JW, United Kingdom We respect your privacy. See https://www.codethink.co.uk/privacy.html

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Only enable with common clock
by Mark Brown 26 Mar '19

26 Mar '19

Some architectures do not yet support the common clock API at all but the tlv320aic32x4 driver now requires it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e99320c79b8..419142111b6d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,8 +180,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER + select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -1110,11 +1110,13 @@ config SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_I2C tristate "Texas Instruments TLV320AIC32x4 audio CODECs - I2C" depends on I2C + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_SPI tristate "Texas Instruments TLV320AIC32x4 audio CODECs - SPI" depends on SPI_MASTER + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC3X -- 2.20.1

1 1

[alsa-devel] Applied "ASoC: Intel: Skylake: enable S24_LE format support" to the asoc tree
by Mark Brown 26 Mar '19

26 Mar '19

The patch ASoC: Intel: Skylake: enable S24_LE format 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 36e075ce74ec4e261a638bf09d10b3348ca4d883 Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Sat, 23 Mar 2019 18:40:10 +0530 Subject: [PATCH] ASoC: Intel: Skylake: enable S24_LE format support To enable S24_LE format, sample_type in topology fw has to be set to 1. But sample_type defined in topology firmware configuration is not getting reflected in the dsp param. This patch sets sample_type in base config so that the sample type defined in the topology firmware is reflected in the dsp params. This issues was uncovered while debugging the S24_LE format which require the MSB byte in 32 bit word to be skipped. Setting sample_type in topology firmware to 1 helps to skip MSB byte word. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Acked-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-messages.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 28c4806b196a..4bf70b4429f0 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -483,6 +483,7 @@ static void skl_set_base_module_format(struct skl_sst *ctx, base_cfg->audio_fmt.bit_depth = format->bit_depth; base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; base_cfg->audio_fmt.ch_cfg = format->ch_cfg; + base_cfg->audio_fmt.sample_type = format->sample_type; dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", format->bit_depth, format->valid_bit_depth, -- 2.20.1

1 0

[alsa-devel] [PATCH v4 0/3] Add NXP AUDMIX device and machine drivers
by Viorel Suman 26 Mar '19

26 Mar '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V3: 1. Removed machine driver DT bindings documentation. 2. Trigger machine driver probe from device driver as suggested by Nicolin. Changes since V2: 1. Moved "dais" node from machine driver DTS node to device driver DTS node as suggested by Rob. Changes since V1: 1. Original patch split into distinct patches for the device driver and DT binding documentation. 2. Replaced AMIX with AUDMIX in both code and file names as it looks more RM-compliant. 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. 4. Added machine driver and related DT binding documentation. Viorel Suman (3): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver .../devicetree/bindings/sound/fsl,audmix.txt | 54 ++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++ sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ 6 files changed, 1080 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c -- 2.7.4

4 7

[alsa-devel] Applied "ASoC: fsl: Add Audio Mixer machine driver" to the asoc tree
by Mark Brown 26 Mar '19

26 Mar '19

The patch ASoC: fsl: Add Audio Mixer machine driver 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 b86ef5367761603df5f66ce08fb014b991f6b51d Mon Sep 17 00:00:00 2001 From: Viorel Suman <viorel.suman(a)nxp.com> Date: Tue, 22 Jan 2019 11:14:30 +0000 Subject: [PATCH] ASoC: fsl: Add Audio Mixer machine driver This patch implements Audio Mixer machine driver for NXP iMX8 SOCs. It connects together Audio Mixer and related SAI instances. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-audmix.c | 327 +++++++++++++++++++++++++++++++++++++ 3 files changed, 338 insertions(+) create mode 100644 sound/soc/fsl/imx-audmix.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0af2e056d211..d87c842806bd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -303,6 +303,15 @@ config SND_SOC_FSL_ASOC_CARD CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. +config SND_SOC_IMX_AUDMIX + tristate "SoC Audio support for i.MX boards with AUDMIX" + select SND_SOC_FSL_AUDMIX + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with Audio Mixer + Say Y if you want to add support for SoC audio on an i.MX board with + an Audio Mixer. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4172d5a3e36c..c0dd04422fe9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -62,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o +snd-soc-imx-audmix-objs := imx-audmix.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o +obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c new file mode 100644 index 000000000000..72e37ca6cfbb --- /dev/null +++ b/sound/soc/fsl/imx-audmix.c @@ -0,0 +1,327 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright 2017 NXP + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/pm_runtime.h> +#include "fsl_sai.h" +#include "fsl_audmix.h" + +struct imx_audmix { + struct platform_device *pdev; + struct snd_soc_card card; + struct platform_device *audmix_pdev; + struct platform_device *out_pdev; + struct clk *cpu_mclk; + int num_dai; + struct snd_soc_dai_link *dai; + int num_dai_conf; + struct snd_soc_codec_conf *dai_conf; + int num_dapm_routes; + struct snd_soc_dapm_route *dapm_routes; +}; + +static const u32 imx_audmix_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = { + .count = ARRAY_SIZE(imx_audmix_rates), + .list = imx_audmix_rates, +}; + +static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->card->dev; + unsigned long clk_rate = clk_get_rate(priv->cpu_mclk); + int ret; + + if (clk_rate % 24576000 == 0) { + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_audmix_rate_constraints); + if (ret < 0) + return ret; + } else { + dev_warn(dev, "mclk may be not supported %lu\n", clk_rate); + } + + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 8); + if (ret < 0) + return ret; + + return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, + FSL_AUDMIX_FORMATS); +} + +static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + u32 channels = params_channels(params); + int ret, dir; + + /* For playback the AUDMIX is slave, and for record is master */ + fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM; + dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + + /* set DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + /* + * Per datasheet, AUDMIX expects 8 slots and 32 bits + * for every slot in TDM mode. + */ + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + BIT(channels) - 1, 8, 32); + if (ret) + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + + return ret; +} + +static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + int ret; + + if (!tx) + return 0; + + /* For playback the AUDMIX is slave */ + fmt |= SND_SOC_DAIFMT_CBM_CFM; + + /* set AUDMIX DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) + dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops imx_audmix_fe_ops = { + .startup = imx_audmix_fe_startup, + .hw_params = imx_audmix_fe_hw_params, +}; + +static struct snd_soc_ops imx_audmix_be_ops = { + .hw_params = imx_audmix_be_hw_params, +}; + +static int imx_audmix_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *audmix_np = NULL, *out_cpu_np = NULL; + struct platform_device *audmix_pdev = NULL; + struct platform_device *cpu_pdev; + struct of_phandle_args args; + struct imx_audmix *priv; + int i, num_dai, ret; + const char *fe_name_pref = "HiFi-AUDMIX-FE-"; + char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name; + + if (pdev->dev.parent) { + audmix_np = pdev->dev.parent->of_node; + } else { + dev_err(&pdev->dev, "Missing parent device.\n"); + return -EINVAL; + } + + if (!audmix_np) { + dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + return -EINVAL; + } + + audmix_pdev = of_find_device_by_node(audmix_np); + if (!audmix_pdev) { + dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n", + np->full_name); + return -EINVAL; + } + + num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); + if (num_dai != FSL_AUDMIX_MAX_DAIS) { + dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n", + audmix_np->full_name); + return -EINVAL; + } + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->num_dai = 2 * num_dai; + priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai * + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + if (!priv->dai) + return -ENOMEM; + + priv->num_dai_conf = num_dai; + priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf * + sizeof(struct snd_soc_codec_conf), + GFP_KERNEL); + if (!priv->dai_conf) + return -ENOMEM; + + priv->num_dapm_routes = 3 * num_dai; + priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes * + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!priv->dapm_routes) + return -ENOMEM; + + for (i = 0; i < num_dai; i++) { + ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, + &args); + if (ret < 0) { + dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); + return ret; + } + + cpu_pdev = of_find_device_by_node(args.np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + + dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", + fe_name_pref, args.np->full_name + 1); + + dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); + + if (i == 0) { + out_cpu_np = args.np; + capture_dai_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Capture"); + } + + priv->dai[i].name = dai_name; + priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; + priv->dai[i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[i].codec_name = "snd-soc-dummy"; + priv->dai[i].cpu_of_node = args.np; + priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platform_of_node = args.np; + priv->dai[i].dynamic = 1; + priv->dai[i].dpcm_playback = 1; + priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); + priv->dai[i].ignore_pmdown_time = 1; + priv->dai[i].ops = &imx_audmix_fe_ops; + + /* Add AUDMIX Backend */ + be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "audmix-%d", i); + be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Playback-%d", i); + be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Capture-%d", i); + + priv->dai[num_dai + i].name = be_name; + priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[num_dai + i].codec_name = "snd-soc-dummy"; + priv->dai[num_dai + i].cpu_of_node = audmix_np; + priv->dai[num_dai + i].cpu_dai_name = be_name; + priv->dai[num_dai + i].platform_name = "snd-soc-dummy"; + priv->dai[num_dai + i].no_pcm = 1; + priv->dai[num_dai + i].dpcm_playback = 1; + priv->dai[num_dai + i].dpcm_capture = 1; + priv->dai[num_dai + i].ignore_pmdown_time = 1; + priv->dai[num_dai + i].ops = &imx_audmix_be_ops; + + priv->dai_conf[i].of_node = args.np; + priv->dai_conf[i].name_prefix = dai_name; + + priv->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Playback"); + priv->dapm_routes[i].sink = be_pb; + priv->dapm_routes[num_dai + i].source = be_pb; + priv->dapm_routes[num_dai + i].sink = be_cp; + priv->dapm_routes[2 * num_dai + i].source = be_cp; + priv->dapm_routes[2 * num_dai + i].sink = capture_dai_name; + } + + cpu_pdev = of_find_device_by_node(out_cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); + if (IS_ERR(priv->cpu_mclk)) { + ret = PTR_ERR(priv->cpu_mclk); + dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); + return -EINVAL; + } + + priv->audmix_pdev = audmix_pdev; + priv->out_pdev = cpu_pdev; + + priv->card.dai_link = priv->dai; + priv->card.num_links = priv->num_dai; + priv->card.codec_conf = priv->dai_conf; + priv->card.num_configs = priv->num_dai_conf; + priv->card.dapm_routes = priv->dapm_routes; + priv->card.num_dapm_routes = priv->num_dapm_routes; + priv->card.dev = pdev->dev.parent; + priv->card.owner = THIS_MODULE; + priv->card.name = "imx-audmix"; + + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + return ret; + } + + return ret; +} + +static struct platform_driver imx_audmix_driver = { + .probe = imx_audmix_probe, + .driver = { + .name = "imx-audmix", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(imx_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman(a)nxp.com>"); +MODULE_ALIAS("platform:imx-audmix"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Only enable with common clock
by Mark Brown 26 Mar '19

26 Mar '19

Some architectures do not yet support the common clock API at all but the tlv320aic32x4 driver now requires it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e99320c79b8..5eafadc855bf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,8 +180,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER + select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C -- 2.20.1

1 0

[alsa-devel] [PATCH v4] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 26 Mar '19

26 Mar '19

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. v3:- o Remove driver supplied time validation. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..eecbfdc7a2c9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.17.1

3 2

Re: [alsa-devel] 'modern dailink' transition
by Kuninori Morimoto 26 Mar '19

26 Mar '19

Oops ?? I exchanged mail address - alsa-devel(a)alsa-devel.org + alsa-devel(a)alsa-project.org > Hi Pierre, Mark > > > > I am however struggling with the notion of a 'snd-soc-dummy' platform that > > > exists in some legacy Intel machine drivers. I changed the code following > > > the pattern below but I have really no idea if this is correct. Shouldn't > > > all dailinks either point to a real platform driver or not provide any > > > information about the platform at all? Is there any specific expectation on > > > the ASoC side here? > > I guess my posted "no Platform" and "no implicit snd-soc-dummy" patch idea > confused you. If so, I'm so sorry about that. > I'm not sure this idea is Good or Bad. > > > I'd expect the dummy driver to just get automatically substituted when > > required, I'd not expect users to explicitly list it. > > Yes agree. > > This is my understanding, please correct me if I was wrong. > I think current many sound card which doesn't need "platfrom" are 2 patterns. > > 1) select snd-soc-dummy as platfrom > 2) select cpu component as platfrom > > Current ASoC selects 1) automatically if .platfrom_name was NULL. > And driver needs to have below if it want to be 2) > > dai_link->platform_of_node = dai_link->cpu_of_node > > But, I think one of them is enough ? > I mean select 2) automatically can be OK? > In other words, current some sound card which doesn't need > platfrom is calling snd-soc-dummy platfrom method in 1) case. > But, is it needed ? I'm not sure... > > It seems snd-soc-dummy platfrom is caring about DPCM-BE case, > but, I think CPU is snd-soc-dummy in such case. > Maybe we need same cade to dummy CPU (?), but *my* DPCM system > is working correctly without it. > > Best regards > --- > Kuninori Morimoto

1 1

[alsa-devel] [asoc:for-5.2 73/82] sound/soc/codecs/tlv320aic32x4-clk.c:258:2: error: unknown field 'prepare' specified in initializer
by kbuild test robot 25 Mar '19

25 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d56ee1550b8a81bc63c80051ff78d8d704b09ba commit: 514b044cba667e4b7c383ec79b42b997e624b91d [73/82] ASoC: tlv320aic32x4: Model PLL in CCF config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 514b044cba667e4b7c383ec79b42b997e624b91d # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/codecs/tlv320aic32x4-clk.c:19:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/init.h:5:0, from include/linux/io.h:22, from include/linux/clk-provider.h:9, from sound/soc/codecs/tlv320aic32x4-clk.c:10: sound/soc/codecs/tlv320aic32x4-clk.c: In function 'clk_aic32x4_pll_prepare': include/linux/kernel.h:979:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:324:9: note: in definition of macro '__compiletime_assert' if (!(condition)) \ ^~~~~~~~~ include/linux/compiler.h:344:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:39:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:979:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:979:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:17:29: note: in expansion of macro 'container_of' #define to_clk_aic32x4(_hw) container_of(_hw, struct clk_aic32x4, hw) ^~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:49:28: note: in expansion of macro 'to_clk_aic32x4' struct clk_aic32x4 *pll = to_clk_aic32x4(hw); ^~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:21: error: variable 'aic32x4_pll_ops' has initializer but incomplete type static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~ >> sound/soc/codecs/tlv320aic32x4-clk.c:258:2: error: unknown field 'prepare' specified in initializer .prepare = clk_aic32x4_pll_prepare, ^ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_pll_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:259:2: error: unknown field 'unprepare' specified in initializer .unprepare = clk_aic32x4_pll_unprepare, ^ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:260:2: error: unknown field 'is_prepared' specified in initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: warning: excess elements in struct initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:261:2: error: unknown field 'recalc_rate' specified in initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:262:2: error: unknown field 'round_rate' specified in initializer .round_rate = clk_aic32x4_pll_round_rate, ^ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:263:2: error: unknown field 'set_rate' specified in initializer .set_rate = clk_aic32x4_pll_set_rate, ^ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:264:2: error: unknown field 'set_parent' specified in initializer .set_parent = clk_aic32x4_pll_set_parent, ^ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:265:2: error: unknown field 'get_parent' specified in initializer .get_parent = clk_aic32x4_pll_get_parent, ^ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:282:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ >> sound/soc/codecs/tlv320aic32x4-clk.c:302:9: error: implicit declaration of function 'devm_clk_register' [-Werror=implicit-function-declaration] return devm_clk_register(dev, &priv->hw); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:282:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:29: error: storage size of 'aic32x4_pll_ops' isn't known static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:303:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/prepare +258 sound/soc/codecs/tlv320aic32x4-clk.c 255 256 257 static const struct clk_ops aic32x4_pll_ops = { > 258 .prepare = clk_aic32x4_pll_prepare, > 259 .unprepare = clk_aic32x4_pll_unprepare, > 260 .is_prepared = clk_aic32x4_pll_is_prepared, > 261 .recalc_rate = clk_aic32x4_pll_recalc_rate, > 262 .round_rate = clk_aic32x4_pll_round_rate, > 263 .set_rate = clk_aic32x4_pll_set_rate, > 264 .set_parent = clk_aic32x4_pll_set_parent, > 265 .get_parent = clk_aic32x4_pll_get_parent, 266 }; 267 268 static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { 269 { 270 .name = "pll", 271 .parent_names = 272 (const char* []) { "mclk", "bclk", "gpio", "din" }, 273 .num_parents = 4, 274 .ops = &aic32x4_pll_ops, 275 .reg = 0, 276 }, 277 }; 278 279 static struct clk *aic32x4_register_clk(struct device *dev, 280 struct aic32x4_clkdesc *desc) 281 { 282 struct clk_init_data init; 283 struct clk_aic32x4 *priv; 284 const char *devname = dev_name(dev); 285 286 init.ops = desc->ops; 287 init.name = desc->name; 288 init.parent_names = desc->parent_names; 289 init.num_parents = desc->num_parents; 290 init.flags = 0; 291 292 priv = devm_kzalloc(dev, sizeof(struct clk_aic32x4), GFP_KERNEL); 293 if (priv == NULL) 294 return (struct clk *) -ENOMEM; 295 296 priv->dev = dev; 297 priv->hw.init = &init; 298 priv->regmap = dev_get_regmap(dev, NULL); 299 priv->reg = desc->reg; 300 301 clk_hw_register_clkdev(&priv->hw, desc->name, devname); > 302 return devm_clk_register(dev, &priv->hw); 303 } 304 --- 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] sound: codecs: fix a potential NULL pointer dereference
by Kangjie Lu 25 Mar '19

25 Mar '19

In case devm_kzalloc fails, the patch returns -ENOMEM to avoid potential NULL pointer dereference. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> --- sound/soc/codecs/rt5663.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index da6647015708..909ab99a1995 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3480,6 +3480,8 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); + if (!rt5663->imp_table) + return -ENOMEM; device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); -- 2.17.1

2 3

[alsa-devel] [asoc:for-5.2 76/82] sound/soc//codecs/tlv320aic32x4-clk.c:372:21: error: variable 'aic32x4_bdiv_ops' has initializer but incomplete type
by kbuild test robot 25 Mar '19

25 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d56ee1550b8a81bc63c80051ff78d8d704b09ba commit: 9b484124ebd906c4d6bc826cc0d417e80cc1105c [76/82] ASoC: tlv320aic32x4: Model BDIV divider in CCF config: sparc64-allyesconfig (attached as .config) compiler: sparc64-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 9b484124ebd906c4d6bc826cc0d417e80cc1105c # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sparc64 All errors (new ones prefixed by >>): .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:259:15: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:260:3: error: 'const struct clk_ops' has no member named 'is_prepared' .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:260:17: warning: excess elements in struct initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:260:17: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:261:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:261:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:261:17: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:262:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:262:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:262:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:263:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:263:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:263:14: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:264:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:264:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:264:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:265:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:265:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:265:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c:287:21: error: variable 'aic32x4_codec_clkin_ops' has initializer but incomplete type static const struct clk_ops aic32x4_codec_clkin_ops = { ^~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:288:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_codec_clkin_set_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:288:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_codec_clkin_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:288:16: note: (near initialization for 'aic32x4_codec_clkin_ops') sound/soc//codecs/tlv320aic32x4-clk.c:289:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_codec_clkin_get_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:289:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_codec_clkin_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:289:16: note: (near initialization for 'aic32x4_codec_clkin_ops') sound/soc//codecs/tlv320aic32x4-clk.c:346:21: error: variable 'aic32x4_div_ops' has initializer but incomplete type static const struct clk_ops aic32x4_div_ops = { ^~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:347:3: error: 'const struct clk_ops' has no member named 'prepare' .prepare = clk_aic32x4_div_prepare, ^~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:347:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_div_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:347:13: note: (near initialization for 'aic32x4_div_ops') sound/soc//codecs/tlv320aic32x4-clk.c:348:3: error: 'const struct clk_ops' has no member named 'unprepare' .unprepare = clk_aic32x4_div_unprepare, ^~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:348:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_div_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:348:15: note: (near initialization for 'aic32x4_div_ops') sound/soc//codecs/tlv320aic32x4-clk.c:349:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_div_set_rate, ^~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:349:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_div_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:349:14: note: (near initialization for 'aic32x4_div_ops') sound/soc//codecs/tlv320aic32x4-clk.c:350:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_div_round_rate, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:350:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_div_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:350:16: note: (near initialization for 'aic32x4_div_ops') sound/soc//codecs/tlv320aic32x4-clk.c:351:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_div_recalc_rate, ^~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:351:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_div_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:351:17: note: (near initialization for 'aic32x4_div_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:372:21: error: variable 'aic32x4_bdiv_ops' has initializer but incomplete type static const struct clk_ops aic32x4_bdiv_ops = { ^~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:373:3: error: 'const struct clk_ops' has no member named 'prepare' .prepare = clk_aic32x4_div_prepare, ^~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:373:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_div_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:373:13: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c:374:3: error: 'const struct clk_ops' has no member named 'unprepare' .unprepare = clk_aic32x4_div_unprepare, ^~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:374:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_div_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:374:15: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c:375:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_bdiv_set_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:375:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_bdiv_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:375:16: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c:376:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_bdiv_get_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:376:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_bdiv_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:376:16: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c:377:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_div_set_rate, ^~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:377:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_div_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:377:14: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c:378:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_div_round_rate, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:378:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_div_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:378:16: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c:379:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_div_recalc_rate, ^~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:379:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_div_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:379:17: note: (near initialization for 'aic32x4_bdiv_ops') sound/soc//codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc//codecs/tlv320aic32x4-clk.c:440:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ sound/soc//codecs/tlv320aic32x4-clk.c:460:9: error: implicit declaration of function 'devm_clk_register'; did you mean 'device_register'? [-Werror=implicit-function-declaration] return devm_clk_register(dev, &priv->hw); ^~~~~~~~~~~~~~~~~ device_register sound/soc//codecs/tlv320aic32x4-clk.c:440:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ sound/soc//codecs/tlv320aic32x4-clk.c: At top level: sound/soc//codecs/tlv320aic32x4-clk.c:257:29: error: storage size of 'aic32x4_pll_ops' isn't known static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:287:29: error: storage size of 'aic32x4_codec_clkin_ops' isn't known static const struct clk_ops aic32x4_codec_clkin_ops = { ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:346:29: error: storage size of 'aic32x4_div_ops' isn't known static const struct clk_ops aic32x4_div_ops = { ^~~~~~~~~~~~~~~ >> sound/soc//codecs/tlv320aic32x4-clk.c:372:29: error: storage size of 'aic32x4_bdiv_ops' isn't known static const struct clk_ops aic32x4_bdiv_ops = { ^~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc//codecs/tlv320aic32x4-clk.c:461:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/aic32x4_bdiv_ops +372 sound/soc//codecs/tlv320aic32x4-clk.c 255 256 > 257 static const struct clk_ops aic32x4_pll_ops = { 258 .prepare = clk_aic32x4_pll_prepare, 259 .unprepare = clk_aic32x4_pll_unprepare, 260 .is_prepared = clk_aic32x4_pll_is_prepared, 261 .recalc_rate = clk_aic32x4_pll_recalc_rate, 262 .round_rate = clk_aic32x4_pll_round_rate, 263 .set_rate = clk_aic32x4_pll_set_rate, 264 .set_parent = clk_aic32x4_pll_set_parent, 265 .get_parent = clk_aic32x4_pll_get_parent, 266 }; 267 268 static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) 269 { 270 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 271 272 return regmap_update_bits(mux->regmap, 273 AIC32X4_CLKMUX, 274 AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); 275 } 276 277 static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) 278 { 279 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 280 unsigned int val; 281 282 regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); 283 284 return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; 285 } 286 287 static const struct clk_ops aic32x4_codec_clkin_ops = { 288 .set_parent = clk_aic32x4_codec_clkin_set_parent, 289 .get_parent = clk_aic32x4_codec_clkin_get_parent, 290 }; 291 292 static int clk_aic32x4_div_prepare(struct clk_hw *hw) 293 { 294 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 295 296 return regmap_update_bits(div->regmap, div->reg, 297 AIC32X4_DIVEN, AIC32X4_DIVEN); 298 } 299 300 static void clk_aic32x4_div_unprepare(struct clk_hw *hw) 301 { 302 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 303 304 regmap_update_bits(div->regmap, div->reg, 305 AIC32X4_DIVEN, 0); 306 } 307 308 static int clk_aic32x4_div_set_rate(struct clk_hw *hw, unsigned long rate, 309 unsigned long parent_rate) 310 { 311 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 312 u8 divisor; 313 314 divisor = DIV_ROUND_UP(parent_rate, rate); 315 if (divisor > 128) 316 return -EINVAL; 317 318 return regmap_update_bits(div->regmap, div->reg, 319 AIC32X4_DIV_MASK, divisor); 320 } 321 322 static long clk_aic32x4_div_round_rate(struct clk_hw *hw, unsigned long rate, 323 unsigned long *parent_rate) 324 { 325 unsigned long divisor; 326 327 divisor = DIV_ROUND_UP(*parent_rate, rate); 328 if (divisor > 128) 329 return -EINVAL; 330 331 return DIV_ROUND_UP(*parent_rate, divisor); 332 } 333 334 static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, 335 unsigned long parent_rate) 336 { 337 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 338 339 unsigned int val; 340 341 regmap_read(div->regmap, div->reg, &val); 342 343 return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); 344 } 345 346 static const struct clk_ops aic32x4_div_ops = { 347 .prepare = clk_aic32x4_div_prepare, 348 .unprepare = clk_aic32x4_div_unprepare, 349 .set_rate = clk_aic32x4_div_set_rate, > 350 .round_rate = clk_aic32x4_div_round_rate, 351 .recalc_rate = clk_aic32x4_div_recalc_rate, 352 }; 353 354 static int clk_aic32x4_bdiv_set_parent(struct clk_hw *hw, u8 index) 355 { 356 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 357 358 return regmap_update_bits(mux->regmap, AIC32X4_IFACE3, 359 AIC32X4_BDIVCLK_MASK, index); 360 } 361 362 static u8 clk_aic32x4_bdiv_get_parent(struct clk_hw *hw) 363 { 364 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 365 unsigned int val; 366 367 regmap_read(mux->regmap, AIC32X4_IFACE3, &val); 368 369 return val & AIC32X4_BDIVCLK_MASK; 370 } 371 > 372 static const struct clk_ops aic32x4_bdiv_ops = { 373 .prepare = clk_aic32x4_div_prepare, 374 .unprepare = clk_aic32x4_div_unprepare, 375 .set_parent = clk_aic32x4_bdiv_set_parent, 376 .get_parent = clk_aic32x4_bdiv_get_parent, 377 .set_rate = clk_aic32x4_div_set_rate, 378 .round_rate = clk_aic32x4_div_round_rate, 379 .recalc_rate = clk_aic32x4_div_recalc_rate, 380 }; 381 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-5.2 75/82] sound/soc/codecs/tlv320aic32x4-clk.c:346:21: error: variable 'aic32x4_div_ops' has initializer but incomplete type
by kbuild test robot 25 Mar '19

25 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d56ee1550b8a81bc63c80051ff78d8d704b09ba commit: a51b50062091619915c5155085bbe13a7aca6903 [75/82] ASoC: tlv320aic32x4: Model DAC/ADC dividers in CCF config: sparc64-allyesconfig (attached as .config) compiler: sparc64-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 a51b50062091619915c5155085bbe13a7aca6903 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sparc64 All errors (new ones prefixed by >>): BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:324:9: note: in definition of macro '__compiletime_assert' if (!(condition)) \ ^~~~~~~~~ include/linux/compiler.h:344:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:39:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:979:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:979:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:17:29: note: in expansion of macro 'container_of' #define to_clk_aic32x4(_hw) container_of(_hw, struct clk_aic32x4, hw) ^~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:49:28: note: in expansion of macro 'to_clk_aic32x4' struct clk_aic32x4 *pll = to_clk_aic32x4(hw); ^~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:21: error: variable 'aic32x4_pll_ops' has initializer but incomplete type static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:3: error: 'const struct clk_ops' has no member named 'prepare' .prepare = clk_aic32x4_pll_prepare, ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_pll_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:259:3: error: 'const struct clk_ops' has no member named 'unprepare' .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:260:3: error: 'const struct clk_ops' has no member named 'is_prepared' .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: warning: excess elements in struct initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:261:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:262:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:263:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:264:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:265:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:287:21: error: variable 'aic32x4_codec_clkin_ops' has initializer but incomplete type static const struct clk_ops aic32x4_codec_clkin_ops = { ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:288:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_codec_clkin_set_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:288:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_codec_clkin_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:288:16: note: (near initialization for 'aic32x4_codec_clkin_ops') sound/soc/codecs/tlv320aic32x4-clk.c:289:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_codec_clkin_get_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:289:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_codec_clkin_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:289:16: note: (near initialization for 'aic32x4_codec_clkin_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:346:21: error: variable 'aic32x4_div_ops' has initializer but incomplete type static const struct clk_ops aic32x4_div_ops = { ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:347:3: error: 'const struct clk_ops' has no member named 'prepare' .prepare = clk_aic32x4_div_prepare, ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:347:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_div_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:347:13: note: (near initialization for 'aic32x4_div_ops') sound/soc/codecs/tlv320aic32x4-clk.c:348:3: error: 'const struct clk_ops' has no member named 'unprepare' .unprepare = clk_aic32x4_div_unprepare, ^~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:348:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_div_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:348:15: note: (near initialization for 'aic32x4_div_ops') sound/soc/codecs/tlv320aic32x4-clk.c:349:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_div_set_rate, ^~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:349:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_div_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:349:14: note: (near initialization for 'aic32x4_div_ops') sound/soc/codecs/tlv320aic32x4-clk.c:350:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_div_round_rate, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:350:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_div_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:350:16: note: (near initialization for 'aic32x4_div_ops') sound/soc/codecs/tlv320aic32x4-clk.c:351:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_div_recalc_rate, ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:351:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_div_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:351:17: note: (near initialization for 'aic32x4_div_ops') sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:404:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ sound/soc/codecs/tlv320aic32x4-clk.c:424:9: error: implicit declaration of function 'devm_clk_register'; did you mean 'device_register'? [-Werror=implicit-function-declaration] return devm_clk_register(dev, &priv->hw); ^~~~~~~~~~~~~~~~~ device_register sound/soc/codecs/tlv320aic32x4-clk.c:404:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:29: error: storage size of 'aic32x4_pll_ops' isn't known static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:287:29: error: storage size of 'aic32x4_codec_clkin_ops' isn't known static const struct clk_ops aic32x4_codec_clkin_ops = { ^~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/codecs/tlv320aic32x4-clk.c:346:29: error: storage size of 'aic32x4_div_ops' isn't known static const struct clk_ops aic32x4_div_ops = { ^~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:425:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/aic32x4_div_ops +346 sound/soc/codecs/tlv320aic32x4-clk.c 255 256 > 257 static const struct clk_ops aic32x4_pll_ops = { 258 .prepare = clk_aic32x4_pll_prepare, 259 .unprepare = clk_aic32x4_pll_unprepare, 260 .is_prepared = clk_aic32x4_pll_is_prepared, 261 .recalc_rate = clk_aic32x4_pll_recalc_rate, 262 .round_rate = clk_aic32x4_pll_round_rate, 263 .set_rate = clk_aic32x4_pll_set_rate, 264 .set_parent = clk_aic32x4_pll_set_parent, 265 .get_parent = clk_aic32x4_pll_get_parent, 266 }; 267 268 static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) 269 { 270 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 271 272 return regmap_update_bits(mux->regmap, 273 AIC32X4_CLKMUX, 274 AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); 275 } 276 277 static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) 278 { 279 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 280 unsigned int val; 281 282 regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); 283 284 return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; 285 } 286 287 static const struct clk_ops aic32x4_codec_clkin_ops = { > 288 .set_parent = clk_aic32x4_codec_clkin_set_parent, 289 .get_parent = clk_aic32x4_codec_clkin_get_parent, 290 }; 291 292 static int clk_aic32x4_div_prepare(struct clk_hw *hw) 293 { 294 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 295 296 return regmap_update_bits(div->regmap, div->reg, 297 AIC32X4_DIVEN, AIC32X4_DIVEN); 298 } 299 300 static void clk_aic32x4_div_unprepare(struct clk_hw *hw) 301 { 302 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 303 304 regmap_update_bits(div->regmap, div->reg, 305 AIC32X4_DIVEN, 0); 306 } 307 308 static int clk_aic32x4_div_set_rate(struct clk_hw *hw, unsigned long rate, 309 unsigned long parent_rate) 310 { 311 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 312 u8 divisor; 313 314 divisor = DIV_ROUND_UP(parent_rate, rate); 315 if (divisor > 128) 316 return -EINVAL; 317 318 return regmap_update_bits(div->regmap, div->reg, 319 AIC32X4_DIV_MASK, divisor); 320 } 321 322 static long clk_aic32x4_div_round_rate(struct clk_hw *hw, unsigned long rate, 323 unsigned long *parent_rate) 324 { 325 unsigned long divisor; 326 327 divisor = DIV_ROUND_UP(*parent_rate, rate); 328 if (divisor > 128) 329 return -EINVAL; 330 331 return DIV_ROUND_UP(*parent_rate, divisor); 332 } 333 334 static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, 335 unsigned long parent_rate) 336 { 337 struct clk_aic32x4 *div = to_clk_aic32x4(hw); 338 339 unsigned int val; 340 341 regmap_read(div->regmap, div->reg, &val); 342 343 return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); 344 } 345 > 346 static const struct clk_ops aic32x4_div_ops = { 347 .prepare = clk_aic32x4_div_prepare, 348 .unprepare = clk_aic32x4_div_unprepare, 349 .set_rate = clk_aic32x4_div_set_rate, 350 .round_rate = clk_aic32x4_div_round_rate, 351 .recalc_rate = clk_aic32x4_div_recalc_rate, 352 }; 353 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-5.2 74/82] sound/soc/codecs/tlv320aic32x4-clk.c:287:21: error: variable 'aic32x4_codec_clkin_ops' has initializer but incomplete type
by kbuild test robot 25 Mar '19

25 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d56ee1550b8a81bc63c80051ff78d8d704b09ba commit: fd2df3aeafa4b4cc468d58e147e0822967034b71 [74/82] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF config: sparc64-allyesconfig (attached as .config) compiler: sparc64-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 fd2df3aeafa4b4cc468d58e147e0822967034b71 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sparc64 All errors (new ones prefixed by >>): sound/soc/codecs/tlv320aic32x4-clk.c:19:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/init.h:5:0, from include/linux/io.h:22, from include/linux/clk-provider.h:9, from sound/soc/codecs/tlv320aic32x4-clk.c:10: sound/soc/codecs/tlv320aic32x4-clk.c: In function 'clk_aic32x4_pll_prepare': include/linux/kernel.h:979:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:324:9: note: in definition of macro '__compiletime_assert' if (!(condition)) \ ^~~~~~~~~ include/linux/compiler.h:344:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:39:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:979:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:979:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:17:29: note: in expansion of macro 'container_of' #define to_clk_aic32x4(_hw) container_of(_hw, struct clk_aic32x4, hw) ^~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:49:28: note: in expansion of macro 'to_clk_aic32x4' struct clk_aic32x4 *pll = to_clk_aic32x4(hw); ^~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:21: error: variable 'aic32x4_pll_ops' has initializer but incomplete type static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:3: error: 'const struct clk_ops' has no member named 'prepare' .prepare = clk_aic32x4_pll_prepare, ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_pll_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:259:3: error: 'const struct clk_ops' has no member named 'unprepare' .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:260:3: error: 'const struct clk_ops' has no member named 'is_prepared' .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: warning: excess elements in struct initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:261:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:262:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:263:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:264:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c:265:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:287:21: error: variable 'aic32x4_codec_clkin_ops' has initializer but incomplete type static const struct clk_ops aic32x4_codec_clkin_ops = { ^~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:288:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_codec_clkin_set_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:288:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_codec_clkin_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:288:16: note: (near initialization for 'aic32x4_codec_clkin_ops') sound/soc/codecs/tlv320aic32x4-clk.c:289:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_codec_clkin_get_parent, ^~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:289:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_codec_clkin_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:289:16: note: (near initialization for 'aic32x4_codec_clkin_ops') sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:314:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ sound/soc/codecs/tlv320aic32x4-clk.c:334:9: error: implicit declaration of function 'devm_clk_register'; did you mean 'device_register'? [-Werror=implicit-function-declaration] return devm_clk_register(dev, &priv->hw); ^~~~~~~~~~~~~~~~~ device_register sound/soc/codecs/tlv320aic32x4-clk.c:314:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:29: error: storage size of 'aic32x4_pll_ops' isn't known static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~~~~~~~~~ >> sound/soc/codecs/tlv320aic32x4-clk.c:287:29: error: storage size of 'aic32x4_codec_clkin_ops' isn't known static const struct clk_ops aic32x4_codec_clkin_ops = { ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:335:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/aic32x4_codec_clkin_ops +287 sound/soc/codecs/tlv320aic32x4-clk.c 255 256 257 static const struct clk_ops aic32x4_pll_ops = { 258 .prepare = clk_aic32x4_pll_prepare, 259 .unprepare = clk_aic32x4_pll_unprepare, 260 .is_prepared = clk_aic32x4_pll_is_prepared, 261 .recalc_rate = clk_aic32x4_pll_recalc_rate, 262 .round_rate = clk_aic32x4_pll_round_rate, 263 .set_rate = clk_aic32x4_pll_set_rate, > 264 .set_parent = clk_aic32x4_pll_set_parent, 265 .get_parent = clk_aic32x4_pll_get_parent, 266 }; 267 268 static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) 269 { 270 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 271 272 return regmap_update_bits(mux->regmap, 273 AIC32X4_CLKMUX, 274 AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); 275 } 276 277 static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) 278 { 279 struct clk_aic32x4 *mux = to_clk_aic32x4(hw); 280 unsigned int val; 281 282 regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); 283 284 return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; 285 } 286 > 287 static const struct clk_ops aic32x4_codec_clkin_ops = { 288 .set_parent = clk_aic32x4_codec_clkin_set_parent, 289 .get_parent = clk_aic32x4_codec_clkin_get_parent, 290 }; 291 292 static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { 293 { 294 .name = "pll", 295 .parent_names = 296 (const char* []) { "mclk", "bclk", "gpio", "din" }, 297 .num_parents = 4, 298 .ops = &aic32x4_pll_ops, 299 .reg = 0, 300 }, 301 { 302 .name = "codec_clkin", 303 .parent_names = 304 (const char *[]) { "mclk", "bclk", "gpio", "pll" }, 305 .num_parents = 4, 306 .ops = &aic32x4_codec_clkin_ops, 307 .reg = 0, 308 }, 309 }; 310 311 static struct clk *aic32x4_register_clk(struct device *dev, 312 struct aic32x4_clkdesc *desc) 313 { > 314 struct clk_init_data init; 315 struct clk_aic32x4 *priv; 316 const char *devname = dev_name(dev); 317 318 init.ops = desc->ops; 319 init.name = desc->name; 320 init.parent_names = desc->parent_names; 321 init.num_parents = desc->num_parents; 322 init.flags = 0; 323 324 priv = devm_kzalloc(dev, sizeof(struct clk_aic32x4), GFP_KERNEL); 325 if (priv == NULL) 326 return (struct clk *) -ENOMEM; 327 328 priv->dev = dev; 329 priv->hw.init = &init; 330 priv->regmap = dev_get_regmap(dev, NULL); 331 priv->reg = desc->reg; 332 333 clk_hw_register_clkdev(&priv->hw, desc->name, devname); 334 return devm_clk_register(dev, &priv->hw); 335 } 336 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:for-5.2 73/82] sound/soc//codecs/tlv320aic32x4-clk.c:257:21: error: variable 'aic32x4_pll_ops' has initializer but incomplete type
by kbuild test robot 25 Mar '19

25 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d56ee1550b8a81bc63c80051ff78d8d704b09ba commit: 514b044cba667e4b7c383ec79b42b997e624b91d [73/82] ASoC: tlv320aic32x4: Model PLL in CCF config: sparc64-allyesconfig (attached as .config) compiler: sparc64-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 514b044cba667e4b7c383ec79b42b997e624b91d # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sparc64 All error/warnings (new ones prefixed by >>): >> sound/soc//codecs/tlv320aic32x4-clk.c:19:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/init.h:5:0, from include/linux/io.h:22, from include/linux/clk-provider.h:9, from sound/soc//codecs/tlv320aic32x4-clk.c:10: sound/soc//codecs/tlv320aic32x4-clk.c: In function 'clk_aic32x4_pll_prepare': include/linux/kernel.h:979:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:324:9: note: in definition of macro '__compiletime_assert' if (!(condition)) \ ^~~~~~~~~ include/linux/compiler.h:344:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:39:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:979:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:979:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ >> sound/soc//codecs/tlv320aic32x4-clk.c:17:29: note: in expansion of macro 'container_of' #define to_clk_aic32x4(_hw) container_of(_hw, struct clk_aic32x4, hw) ^~~~~~~~~~~~ >> sound/soc//codecs/tlv320aic32x4-clk.c:49:28: note: in expansion of macro 'to_clk_aic32x4' struct clk_aic32x4 *pll = to_clk_aic32x4(hw); ^~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c: At top level: >> sound/soc//codecs/tlv320aic32x4-clk.c:257:21: error: variable 'aic32x4_pll_ops' has initializer but incomplete type static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~ >> sound/soc//codecs/tlv320aic32x4-clk.c:258:3: error: 'const struct clk_ops' has no member named 'prepare' .prepare = clk_aic32x4_pll_prepare, ^~~~~~~ >> sound/soc//codecs/tlv320aic32x4-clk.c:258:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_pll_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:258:13: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:259:3: error: 'const struct clk_ops' has no member named 'unprepare' .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:259:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:259:15: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:260:3: error: 'const struct clk_ops' has no member named 'is_prepared' .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:260:17: warning: excess elements in struct initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:260:17: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:261:3: error: 'const struct clk_ops' has no member named 'recalc_rate' .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:261:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:261:17: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:262:3: error: 'const struct clk_ops' has no member named 'round_rate' .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:262:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:262:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:263:3: error: 'const struct clk_ops' has no member named 'set_rate' .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:263:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:263:14: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:264:3: error: 'const struct clk_ops' has no member named 'set_parent' .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:264:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:264:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc//codecs/tlv320aic32x4-clk.c:265:3: error: 'const struct clk_ops' has no member named 'get_parent' .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:265:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c:265:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc//codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': >> sound/soc//codecs/tlv320aic32x4-clk.c:282:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ >> sound/soc//codecs/tlv320aic32x4-clk.c:302:9: error: implicit declaration of function 'devm_clk_register'; did you mean 'device_register'? [-Werror=implicit-function-declaration] return devm_clk_register(dev, &priv->hw); ^~~~~~~~~~~~~~~~~ device_register sound/soc//codecs/tlv320aic32x4-clk.c:282:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ sound/soc//codecs/tlv320aic32x4-clk.c: At top level: >> sound/soc//codecs/tlv320aic32x4-clk.c:257:29: error: storage size of 'aic32x4_pll_ops' isn't known static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~~~~~~~~~ sound/soc//codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': >> sound/soc//codecs/tlv320aic32x4-clk.c:303:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/aic32x4_pll_ops +257 sound/soc//codecs/tlv320aic32x4-clk.c 255 256 > 257 static const struct clk_ops aic32x4_pll_ops = { > 258 .prepare = clk_aic32x4_pll_prepare, > 259 .unprepare = clk_aic32x4_pll_unprepare, > 260 .is_prepared = clk_aic32x4_pll_is_prepared, > 261 .recalc_rate = clk_aic32x4_pll_recalc_rate, > 262 .round_rate = clk_aic32x4_pll_round_rate, > 263 .set_rate = clk_aic32x4_pll_set_rate, > 264 .set_parent = clk_aic32x4_pll_set_parent, > 265 .get_parent = clk_aic32x4_pll_get_parent, 266 }; 267 268 static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { 269 { 270 .name = "pll", 271 .parent_names = 272 (const char* []) { "mclk", "bclk", "gpio", "din" }, 273 .num_parents = 4, 274 .ops = &aic32x4_pll_ops, 275 .reg = 0, 276 }, 277 }; 278 279 static struct clk *aic32x4_register_clk(struct device *dev, 280 struct aic32x4_clkdesc *desc) 281 { > 282 struct clk_init_data init; 283 struct clk_aic32x4 *priv; 284 const char *devname = dev_name(dev); 285 286 init.ops = desc->ops; 287 init.name = desc->name; 288 init.parent_names = desc->parent_names; 289 init.num_parents = desc->num_parents; 290 init.flags = 0; 291 292 priv = devm_kzalloc(dev, sizeof(struct clk_aic32x4), GFP_KERNEL); 293 if (priv == NULL) 294 return (struct clk *) -ENOMEM; 295 296 priv->dev = dev; 297 priv->hw.init = &init; 298 priv->regmap = dev_get_regmap(dev, NULL); 299 priv->reg = desc->reg; 300 301 clk_hw_register_clkdev(&priv->hw, desc->name, devname); > 302 return devm_clk_register(dev, &priv->hw); > 303 } 304 --- 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: intel: Fix crash at suspend/resume after failed codec registration
by Guenter Roeck 25 Mar '19

25 Mar '19

If codec registration fails after the ASoC Intel SST driver has been probed, the kernel will Oops and crash at suspend/resume. general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI CPU: 1 PID: 2811 Comm: cat Tainted: G W 4.19.30 #15 Hardware name: GOOGLE Clapper, BIOS Google_Clapper.5216.199.7 08/22/2014 RIP: 0010:snd_soc_suspend+0x5a/0xd21 Code: 03 80 3c 10 00 49 89 d7 74 0b 48 89 df e8 71 72 c4 fe 4c 89 fa 48 8b 03 48 89 45 d0 48 8d 98 a0 01 00 00 48 89 d8 48 c1 e8 03 <8a> 04 10 84 c0 0f 85 85 0c 00 00 80 3b 00 0f 84 6b 0c 00 00 48 8b RSP: 0018:ffff888035407750 EFLAGS: 00010202 RAX: 0000000000000034 RBX: 00000000000001a0 RCX: 0000000000000000 RDX: dffffc0000000000 RSI: 0000000000000008 RDI: ffff88805c417098 RBP: ffff8880354077b0 R08: dffffc0000000000 R09: ffffed100b975718 R10: 0000000000000001 R11: ffffffff949ea4a3 R12: 1ffff1100b975746 R13: dffffc0000000000 R14: ffff88805cba4588 R15: dffffc0000000000 FS: 0000794a78e91b80(0000) GS:ffff888068d00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007bd5283ccf58 CR3: 000000004b7aa000 CR4: 00000000001006e0 Call Trace: ? dpm_complete+0x67b/0x67b ? i915_gem_suspend+0x14d/0x1ad sst_soc_prepare+0x91/0x1dd ? sst_be_hw_params+0x7e/0x7e dpm_prepare+0x39a/0x88b dpm_suspend_start+0x13/0x9d suspend_devices_and_enter+0x18f/0xbd7 ? arch_suspend_enable_irqs+0x11/0x11 ? printk+0xd9/0x12d ? lock_release+0x95f/0x95f ? log_buf_vmcoreinfo_setup+0x131/0x131 ? rcu_read_lock_sched_held+0x140/0x22a ? __bpf_trace_rcu_utilization+0xa/0xa ? __pm_pr_dbg+0x186/0x190 ? pm_notifier_call_chain+0x39/0x39 ? suspend_test+0x9d/0x9d pm_suspend+0x2f4/0x728 ? trace_suspend_resume+0x3da/0x3da ? lock_release+0x95f/0x95f ? kernfs_fop_write+0x19f/0x32d state_store+0xd8/0x147 ? sysfs_kf_read+0x155/0x155 kernfs_fop_write+0x23e/0x32d __vfs_write+0x108/0x608 ? vfs_read+0x2e9/0x2e9 ? rcu_read_lock_sched_held+0x140/0x22a ? __bpf_trace_rcu_utilization+0xa/0xa ? debug_smp_processor_id+0x10/0x10 ? selinux_file_permission+0x1c5/0x3c8 ? rcu_sync_lockdep_assert+0x6a/0xad ? __sb_start_write+0x129/0x2ac vfs_write+0x1aa/0x434 ksys_write+0xfe/0x1be ? __ia32_sys_read+0x82/0x82 do_syscall_64+0xcd/0x120 entry_SYSCALL_64_after_hwframe+0x49/0xbe In the observed situation, the problem is seen because the codec driver failed to probe due to a hardware problem. max98090 i2c-193C9890:00: Failed to read device revision: -1 max98090 i2c-193C9890:00: ASoC: failed to probe component -1 cht-bsw-max98090 cht-bsw-max98090: ASoC: failed to instantiate card -1 cht-bsw-max98090 cht-bsw-max98090: snd_soc_register_card failed -1 cht-bsw-max98090: probe of cht-bsw-max98090 failed with error -1 The problem is similar to the problem solved with commit 2fc995a87f2e ("ASoC: intel: Fix crash at suspend/resume without card registration"), but codec registration fails at a later point. At that time, the pointer checked with the above mentioned commit is already set, but it is not cleared if the device is subsequently removed. Adding a remove function to clear the pointer fixes the problem. Cc: stable(a)vger.kernel.org Cc: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 08cea5b5cda9..0e8b1c5eec88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -706,9 +706,17 @@ static int sst_soc_probe(struct snd_soc_component *component) return sst_dsp_init_v2_dpcm(component); } +static void sst_soc_remove(struct snd_soc_component *component) +{ + struct sst_data *drv = dev_get_drvdata(component->dev); + + drv->soc_card = NULL; +} + static const struct snd_soc_component_driver sst_soc_platform_drv = { .name = DRV_NAME, .probe = sst_soc_probe, + .remove = sst_soc_remove, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, -- 2.7.4

3 9

[alsa-devel] [PATCH] ASoC: dapm: Initialize private_value in snd_soc_dapm_new_dai
by Pankaj Bharadiya 25 Mar '19

25 Mar '19

In case of single config, private_value is left uninitialized. The private_value does need to be initialized or in snd_soc_dapm_new_control_unlocked() call failure case, it leads to a bogus free in snd_soc_dapm_free_kcontrol() Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya(a)intel.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 67b032c..665a6b1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4042,7 +4042,7 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; const char **w_param_text; - unsigned long private_value; + unsigned long private_value = 0; char *link_name; int ret; -- 2.7.4

2 1

[alsa-devel] [PATCH v4 00/10] ASoC: tlv320aic32x4: Rework Clock Setting
by Annaliese McDermond 25 Mar '19

25 Mar '19

The current tlv320aic32x4 code sets the various clock parameters by looking in a static table in the kernel. This works well enough, but it has a few disadvantages. 1) If the master clock doesn't match one of the precalculated values the driver cannot work. This could be because the designer decided to use a different crystal, or because the clock comes from a PLL that can't quite achieve the proper frequency exactly. 2) The driver only supports certain pre-calculated sample rates. The actual hardware can support many more. These changes enable those rates. Additionally, certain sample rates at certain clock rates were previously unavailable as combinations. This patch dynamically calculates the various PLL and divider values to find something that works. Additionally, to enable much of this, the clock tree is modeled as a part of the Common Clock Framework. This allows for easier debugigng as you can use the standard tools such as /sys/kernel/debug/clk/clk_summary. It also simplifies enabling the entirety of the clock tree needed, and makes sure unused clocks become disabled. These patches have been tested and are working on a Raspberry Pi with the simplecard driver. Changes in v2: - Update 0002 patch to change Kconfig to add dependency on CCF - Update 0002 patch to correct the SPDX identifier - Update 0002 patch to use EXPORT_SYMBOL_GPL() instead of EXPORT_SYMBOL() - Update patches 0002, 0004 and 0005 to use prepare/unprepare instead of enable/disable since the i2c operations are not atomic - Update 0004 patch to fix indentation in commit message - Drop patch 0009 because it is a duplicate of the already committed patch in 667e9334 - Update 0019 patch to keep aic32x4_set_dai_sysclk but to have it use clk_set_rate to set the master clock rate Changes in v3: - Rebase on current for-next - Move "Properly Set Processing Blocks" to the start of the patches so that it can potentially be applied as a bug fix Changes in v4: - Rabase on current for-next - Regenerate patches to get them to apply correctly. Verified with git-am on for-next and they appear to work here. Annaliese McDermond (10): ASoC: tlv320aic32x4: Model PLL in CCF ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF ASoC: tlv320aic32x4: Model DAC/ADC dividers in CCF ASoC: tlv320aic32x4: Model BDIV divider in CCF ASoC: tlv320aic32x4: Control clock gating with CCF ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions ASoC: tlv320aic32x4: Dynamically Determine Clocking ASoC: tlv320aic32x4: Restructure set_dai_sysclk ASoC: tlv320aic32x4: Remove mclk references ASoC: tlv320aic32x4: Allow 192000 Sample Rate sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tlv320aic32x4-clk.c | 483 +++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 427 +++++++++++------------ sound/soc/codecs/tlv320aic32x4.h | 11 + 5 files changed, 692 insertions(+), 232 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4-clk.c -- 2.19.1

2 13

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF 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 fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model BDIV divider in CCF" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Model BDIV divider in CCF 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 9b484124ebd906c4d6bc826cc0d417e80cc1105c Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:48 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model BDIV divider in CCF Model and manage BDIV divider as components in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 36 ++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 56 +++++++++++++--------------- 2 files changed, 62 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index daf14924e324..667ec2c03508 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -351,6 +351,34 @@ static const struct clk_ops aic32x4_div_ops = { .recalc_rate = clk_aic32x4_div_recalc_rate, }; +static int clk_aic32x4_bdiv_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, index); +} + +static u8 clk_aic32x4_bdiv_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_IFACE3, &val); + + return val & AIC32X4_BDIVCLK_MASK; +} + +static const struct clk_ops aic32x4_bdiv_ops = { + .prepare = clk_aic32x4_div_prepare, + .unprepare = clk_aic32x4_div_unprepare, + .set_parent = clk_aic32x4_bdiv_set_parent, + .get_parent = clk_aic32x4_bdiv_get_parent, + .set_rate = clk_aic32x4_div_set_rate, + .round_rate = clk_aic32x4_div_round_rate, + .recalc_rate = clk_aic32x4_div_recalc_rate, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -396,6 +424,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_div_ops, .reg = AIC32X4_MADC, }, + { + .name = "bdiv", + .parent_names = + (const char *[]) { "ndac", "mdac", "nadc", "madc" }, + .num_parents = 4, + .ops = &aic32x4_bdiv_ops, + .reg = AIC32X4_BCLKN, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 0cf942938372..bf9ccda6616d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -57,7 +57,7 @@ struct aic32x4_rate_divs { u8 aosr; unsigned long nadc_rate; unsigned long madc_rate; - u8 blck_N; + unsigned long bdiv_rate; u8 r_block; u8 p_block; }; @@ -310,53 +310,53 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 8k rate */ { 12000000, 8000, 57120000, 768, 18432000, 6144000, 128, 18432000, - 1024000, 24, 1, 1 }, + 1024000, 256000, 1, 1 }, { 24000000, 8000, 57120000, 768, 6144000, 6144000, 64, 2048000, - 512000, 24, 1, 1 }, + 512000, 256000, 1, 1 }, { 25000000, 8000, 32620000, 768, 6144000, 6144000, 64, 2048000, - 512000, 24, 1, 1 }, + 512000, 256000, 1, 1 }, /* 11.025k rate */ { 12000000, 11025, 44217600, 512, 11289600, 5644800, 128, 11289600, - 1411200, 16, 1, 1 }, + 1411200, 352800, 1, 1 }, { 24000000, 11025, 44217600, 512, 5644800, 5644800, 64, 2822400, - 705600, 16, 1, 1 }, + 705600, 352800, 1, 1 }, /* 16k rate */ { 12000000, 16000, 57120000, 384, 18432000, 6144000, 128, 18432000, - 2048000, 12, 1, 1 }, + 2048000, 512000, 1, 1 }, { 24000000, 16000, 57120000, 384, 6144000, 6144000, 64, 5120000, - 1024000, 12, 1, 1 }, + 1024000, 512000, 1, 1 }, { 25000000, 16000, 32620000, 384, 6144000, 6144000, 64, 5120000, - 1024000, 12, 1, 1 }, + 1024000, 512000, 1, 1 }, /* 22.05k rate */ { 12000000, 22050, 44217600, 256, 22579200, 5644800, 128, 22579200, - 2822400, 8, 1, 1 }, + 2822400, 705600, 1, 1 }, { 24000000, 22050, 44217600, 256, 5644800, 5644800, 64, 5644800, - 1411200, 8, 1, 1 }, + 1411200, 705600, 1, 1 }, { 25000000, 22050, 19713750, 256, 5644800, 5644800, 64, 5644800, - 1411200, 8, 1, 1 }, + 1411200, 705600, 1, 1 }, /* 32k rate */ { 12000000, 32000, 14112000, 192, 43008000, 6144000, 64, 43008000, - 2048000, 6, 1, 1 }, + 2048000, 1024000, 1, 1 }, { 24000000, 32000, 14112000, 192, 12288000, 6144000, 64, 12288000, - 2048000, 6, 1, 1 }, + 2048000, 1024000, 1, 1 }, /* 44.1k rate */ { 12000000, 44100, 44217600, 128, 45158400, 5644800, 128, 45158400, - 5644800, 4, 1, 1 }, + 5644800, 1411200, 1, 1 }, { 24000000, 44100, 44217600, 128, 11289600, 5644800, 64, 11289600, - 2822400, 4, 1, 1 }, + 2822400, 1411200, 1, 1 }, { 25000000, 44100, 19713750, 128, 11289600, 5644800, 64, 11289600, - 2822400, 4, 1, 1 }, + 2822400, 1411200, 1, 1 }, /* 48k rate */ { 12000000, 48000, 18432000, 128, 49152000, 6144000, 128, 49152000, - 6144000, 4, 1, 1 }, + 6144000, 1536000, 1, 1 }, { 24000000, 48000, 18432000, 128, 12288000, 6144000, 64, 12288000, - 3072000, 4, 1, 1 }, + 3072000, 1536000, 1, 1 }, { 25000000, 48000, 75626250, 128, 12288000, 6144000, 64, 12288000, - 3072000, 4, 1, 1 }, + 3072000, 1536000, 1, 1 }, /* 96k rate */ { 25000000, 96000, 75626250, 64, 24576000, 6144000, 64, 24576000, - 6144000, 1, 1, 9 }, + 6144000, 3072000, 1, 9 }, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -743,6 +743,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, { .id = "madc" }, { .id = "ndac" }, { .id = "mdac" }, + { .id = "bdiv" }, }; i = aic32x4_get_divs(parent_rate, sample_rate); @@ -760,14 +761,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, clk_set_rate(clocks[2].clk, aic32x4_divs[i].madc_rate); clk_set_rate(clocks[3].clk, aic32x4_divs[i].ndac_rate); clk_set_rate(clocks[4].clk, aic32x4_divs[i].mdac_rate); + clk_set_rate(clocks[5].clk, aic32x4_divs[i].bdiv_rate); aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, - AIC32X4_BDIVCLK_MASK, - AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); - /* DOSR MSB & LSB values */ snd_soc_component_write(component, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); snd_soc_component_write(component, AIC32X4_DOSRLSB, (aic32x4_divs[i].dosr & 0xff)); @@ -775,10 +772,6 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, /* AOSR value */ snd_soc_component_write(component, AIC32X4_AOSR, aic32x4_divs[i].aosr); - /* BCLK N divider */ - snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLK_MASK, aic32x4_divs[i].blck_N); - return 0; } @@ -1001,6 +994,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) struct clk_bulk_data clocks[] = { { .id = "codec_clkin" }, { .id = "pll" }, + { .id = "bdiv" }, + { .id = "mdac" }, }; ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); @@ -1019,6 +1014,7 @@ static int aic32x4_component_probe(struct snd_soc_component *component) aic32x4_setup_gpios(component); clk_set_parent(clocks[0].clk, clocks[1].clk); + clk_set_parent(clocks[2].clk, clocks[3].clk); /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Control clock gating with CCF" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Control clock gating with CCF 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 d25970b5fd51e9fcf0afbe190908ea4049454da4 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:49 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Control clock gating with CCF Control the clock gating to the various clock components to use the CCF. This allows us to prepare_enalbe only 3 clocks and the relationships assigned to them will cause upstream clockss to enable automatically. Additionally we can do this in a single call to the CCF. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 67 +++++++------------------------- 1 file changed, 13 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index bf9ccda6616d..c3e9f65b26a7 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -834,41 +834,25 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) static int aic32x4_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { - struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); int ret; + struct clk_bulk_data clocks[] = { + { .id = "madc" }, + { .id = "mdac" }, + { .id = "bdiv" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; + switch (level) { case SND_SOC_BIAS_ON: - /* Switch on master clock */ - ret = clk_prepare_enable(aic32x4->mclk); + ret = clk_bulk_prepare_enable(ARRAY_SIZE(clocks), clocks); if (ret) { - dev_err(component->dev, "Failed to enable master clock\n"); + dev_err(component->dev, "Failed to enable clocks\n"); return ret; } - - /* Switch on PLL */ - snd_soc_component_update_bits(component, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); - - /* Switch on NDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); - - /* Switch on MDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); - - /* Switch on NADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); - - /* Switch on MADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); - - /* Switch on BCLK_N Divider */ - snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; @@ -877,32 +861,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) break; - /* Switch off BCLK_N Divider */ - snd_soc_component_update_bits(component, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); - - /* Switch off MADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MADC, - AIC32X4_MADCEN, 0); - - /* Switch off NADC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NADC, - AIC32X4_NADCEN, 0); - - /* Switch off MDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); - - /* Switch off NDAC Divider */ - snd_soc_component_update_bits(component, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); - - /* Switch off PLL */ - snd_soc_component_update_bits(component, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); - - /* Switch off master clock */ - clk_disable_unprepare(aic32x4->mclk); + clk_bulk_disable_unprepare(ARRAY_SIZE(clocks), clocks); break; case SND_SOC_BIAS_OFF: break; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions 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 fbafbf6517274a797e6e6508c18dd8dba5920c89 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:50 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions Move these to separate helper functions. This looks cleaner and fits better with the new clock setting in CCF. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c3e9f65b26a7..bf3a80237b03 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -718,6 +718,20 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int aic32x4_set_aosr(struct snd_soc_component *component, u8 aosr) +{ + return snd_soc_component_write(component, AIC32X4_AOSR, aosr); +} + +static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr) +{ + snd_soc_component_write(component, AIC32X4_DOSRMSB, dosr >> 8); + snd_soc_component_write(component, AIC32X4_DOSRLSB, + (dosr & 0xff)); + + return 0; +} + static int aic32x4_set_processing_blocks(struct snd_soc_component *component, u8 r_block, u8 p_block) { @@ -763,14 +777,10 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, clk_set_rate(clocks[4].clk, aic32x4_divs[i].mdac_rate); clk_set_rate(clocks[5].clk, aic32x4_divs[i].bdiv_rate); - aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); + aic32x4_set_aosr(component, aic32x4_divs[i].aosr); + aic32x4_set_dosr(component, aic32x4_divs[i].dosr); - /* DOSR MSB & LSB values */ - snd_soc_component_write(component, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); - snd_soc_component_write(component, AIC32X4_DOSRLSB, (aic32x4_divs[i].dosr & 0xff)); - - /* AOSR value */ - snd_soc_component_write(component, AIC32X4_AOSR, aic32x4_divs[i].aosr); + aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); return 0; } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Restructure set_dai_sysclk" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Restructure set_dai_sysclk 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 aa6a60f7be925210d5156f0e8025f3afe1f4f54d Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:52 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Restructure set_dai_sysclk The sysclk is now managed by the CCF. Change this function to merely find the system clock and set it using clk_set_rate. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 625528500f8d..289bf411e60e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -49,7 +49,6 @@ struct aic32x4_priv { struct regmap *regmap; - u32 sysclk; u32 power_cfg; u32 micpga_routing; bool swapdacs; @@ -567,17 +566,13 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_component *component = codec_dai->component; - struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + struct clk *mclk; + struct clk *pll; - switch (freq) { - case 12000000: - case 24000000: - case 25000000: - aic32x4->sysclk = freq; - return 0; - } - printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); - return -EINVAL; + pll = devm_clk_get(component->dev, "pll"); + mclk = clk_get_parent(pll); + + return clk_set_rate(mclk, freq); } static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Remove mclk references" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Remove mclk references 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 78f2d58a289302e56a7def96a783a7686ebf27e2 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:53 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Remove mclk references mclk is not used by anything anymore. Remove support for it. All that information now comes from the clock tree. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 289bf411e60e..a5457d3de1b1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -53,7 +53,6 @@ struct aic32x4_priv { u32 micpga_routing; bool swapdacs; int rstn_gpio; - struct clk *mclk; const char *mclk_name; struct regulator *supply_ldo; @@ -1190,12 +1189,6 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) aic32x4->mclk_name = "mclk"; } - aic32x4->mclk = devm_clk_get(dev, "mclk"); - if (IS_ERR(aic32x4->mclk)) { - dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); - return PTR_ERR(aic32x4->mclk); - } - ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); if (ret) return ret; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Allow 192000 Sample Rate" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: tlv320aic32x4: Allow 192000 Sample Rate 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 6d56ee1550b8a81bc63c80051ff78d8d704b09ba Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:54 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Allow 192000 Sample Rate The clocking and processing blocks are now properly set up to support 192000 sample rates. Allow drivers to ask for that. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index a5457d3de1b1..75443efeda69 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -857,7 +857,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_192000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -- 2.20.1

1 0

[alsa-devel] [PATCH] ALSA: aloop: Support S24 sample formats
by twischer＠de.adit-jv.com 25 Mar '19

25 Mar '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Currently snd_aloop supports only S16 and S32 audio sample formats. With this patch the S24 formats are also supported. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/drivers/aloop.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 8c3fbe1..c14e57b 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -337,7 +337,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) loopback_timer_stop_sync(dpcm); - salign = (snd_pcm_format_width(runtime->format) * + salign = (snd_pcm_format_physical_width(runtime->format) * runtime->channels) / 8; bps = salign * runtime->rate; if (bps <= 0 || salign <= 0) @@ -562,6 +562,8 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, -- 2.7.4

3 2

[alsa-devel] [PROBLEM] ALSA: hda: PCM streams are suspended while opening
by Jon Hunter 25 Mar '19

25 Mar '19

From: Jonathan Hunter <jonathanh(a)nvidia.com> This issue is not observed on the latest mainline kernel since the 'ALSA: PCM suspend cleanup' series has been applied and the snd_pcm_suspend_all() function call in the HDA codec driver has been removed. However, this issue impacts stable kernel branches and so I am trying to find a solution for these branches. When stress testing audio playback across multiple HDA streams simultaneously, the following failure is sometimes observed when starting playback ... Unable to set hw params for playback: File descriptor in bad state The problem is caused because ... 1. Playback is starting for one HDA codec and so the chip->open_mutex in azx_pcm_open() has been acquired. 2. Playback for another HDA codec is starting but is blocked waiting to acquire the chip->open_mutex in azx_pcm_open(). 3. For the HDA codec that is blocked, its runtime-pm status is still enabled from a previous playback session that has since completed and been closed, however its autosuspend timeout has not expired yet. 4. For the HDA codec that is blocked, the runtime-pm autosuspend timeout now occurs calling the runtime-pm suspend callback and this calls snd_pcm_suspend_all() placing all PCM streams in the suspended state. 5. The block HDA codec is now unblocked and fails to set the HW params because the PCM stream is in the suspended state. The above has been observed on Tegra platforms where the autosuspend delay is set to 1 second and there is a delay to an AZX command. This highlights that there is a window of time where autosuspend can place the HDA PCM stream in the suspended state while opening the stream and cause playback to fail. Looking at commit 3d21ef0b49f8 ('ALSA: pcm: Suspend streams globally via device type PM ops') it appears that the call to snd_pcm_suspend_all() has now been moved to the to the suspend handler for PCM streams and so I am wondering if it would be equivalent to make the following change to the HDA codec driver to achieve the same behaviour but only for the HDA driver. So far the issue has not been observed with this change. Please note that I am not sending this as a formal patch as I wanted to get some feedback/comments on the above. --- sound/pci/hda/hda_codec.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9f8d59e7e89f..26ecc6e1f524 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2923,12 +2923,9 @@ static void hda_call_codec_resume(struct hda_codec *codec) static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - struct hda_pcm *pcm; unsigned int state; cancel_delayed_work_sync(&codec->jackpoll_work); - list_for_each_entry(pcm, &codec->pcm_list_head, list) - snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); if (codec->link_down_at_suspend || (codec_has_clkstop(codec) && codec_has_epss(codec) && @@ -2948,11 +2945,22 @@ static int hda_codec_runtime_resume(struct device *dev) pm_runtime_mark_last_busy(dev); return 0; } + +static int hda_codec_suspend(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + struct hda_pcm *pcm; + + list_for_each_entry(pcm, &codec->pcm_list_head, list) + snd_pcm_suspend_all(pcm->pcm); + + return pm_runtime_force_suspend(dev); +} #endif /* CONFIG_PM */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + SET_SYSTEM_SLEEP_PM_OPS(hda_codec_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, NULL) -- 1.9.1

2 8

[alsa-devel] [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit
by John Hsu 25 Mar '19

25 Mar '19

Add suffix ULL to constant 256 in order to give the compiler complete information about the proper arithmetic to use. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index bfd74b86c9d2..e7fd0b2645ea 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -505,7 +505,7 @@ static int nau88l0_calc_pll(unsigned int pll_in, f2_max = 0; scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { - f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- 2.18.0

2 2

[alsa-devel] Applied "ASoC: AMD: Fix incorrect extern" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: AMD: Fix incorrect extern 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 eddb6084305e5418ce28b8379a6c3d0058a61463 Mon Sep 17 00:00:00 2001 From: Andi Kleen <ak(a)linux.intel.com> Date: Thu, 21 Mar 2019 15:00:05 -0700 Subject: [PATCH] ASoC: AMD: Fix incorrect extern When using bare externs outside include files that types should at least match. This fixes a type confusion between bool and int. Cc: broonie(a)kernel.org Signed-off-by: Andi Kleen <ak(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- 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 f37a588ba345..8a619a75b3a9 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,7 +47,7 @@ static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; -extern int bt_uart_enable; +extern bool bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: stm32: sai: add power management
by Olivier Moysan 25 Mar '19

25 Mar '19

Add support of low power modes to STM32 SAI driver. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai.c | 80 ++++++++++++++++++++++++++++++++++++++----- sound/soc/stm/stm32_sai.h | 2 ++ sound/soc/stm/stm32_sai_sub.c | 45 ++++++++++++++++++------ 3 files changed, 108 insertions(+), 19 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index d68d62f12df5..7550d5f08be3 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_platform.h> +#include <linux/pinctrl/consumer.h> #include <linux/reset.h> #include <sound/dmaengine_pcm.h> @@ -44,20 +45,41 @@ static const struct of_device_id stm32_sai_ids[] = { {} }; -static int stm32_sai_sync_conf_client(struct stm32_sai_data *sai, int synci) +static int stm32_sai_pclk_disable(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + + clk_disable_unprepare(sai->pclk); + + return 0; +} + +static int stm32_sai_pclk_enable(struct device *dev) { + struct stm32_sai_data *sai = dev_get_drvdata(dev); int ret; - /* Enable peripheral clock to allow GCR register access */ ret = clk_prepare_enable(sai->pclk); if (ret) { dev_err(&sai->pdev->dev, "failed to enable clock: %d\n", ret); return ret; } + return 0; +} + +static int stm32_sai_sync_conf_client(struct stm32_sai_data *sai, int synci) +{ + int ret; + + /* Enable peripheral clock to allow GCR register access */ + ret = stm32_sai_pclk_enable(&sai->pdev->dev); + if (ret) + return ret; + writel_relaxed(FIELD_PREP(SAI_GCR_SYNCIN_MASK, (synci - 1)), sai->base); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return 0; } @@ -68,11 +90,9 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) int ret; /* Enable peripheral clock to allow GCR register access */ - ret = clk_prepare_enable(sai->pclk); - if (ret) { - dev_err(&sai->pdev->dev, "failed to enable clock: %d\n", ret); + ret = stm32_sai_pclk_enable(&sai->pdev->dev); + if (ret) return ret; - } dev_dbg(&sai->pdev->dev, "Set %pOFn%s as synchro provider\n", sai->pdev->dev.of_node, @@ -83,13 +103,13 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } writel_relaxed(FIELD_PREP(SAI_GCR_SYNCOUT_MASK, synco), sai->base); - clk_disable_unprepare(sai->pclk); + stm32_sai_pclk_disable(&sai->pdev->dev); return 0; } @@ -195,12 +215,54 @@ static int stm32_sai_probe(struct platform_device *pdev) return devm_of_platform_populate(&pdev->dev); } +#ifdef CONFIG_PM_SLEEP +/* + * When pins are shared by two sai sub instances, pins have to be defined + * in sai parent node. In this case, pins state is not managed by alsa fw. + * These pins are managed in suspend/resume callbacks. + */ +static int stm32_sai_suspend(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + int ret; + + ret = stm32_sai_pclk_enable(dev); + if (ret) + return ret; + + sai->gcr = readl_relaxed(sai->base); + stm32_sai_pclk_disable(dev); + + return pinctrl_pm_select_sleep_state(dev); +} + +static int stm32_sai_resume(struct device *dev) +{ + struct stm32_sai_data *sai = dev_get_drvdata(dev); + int ret; + + ret = stm32_sai_pclk_enable(dev); + if (ret) + return ret; + + writel_relaxed(sai->gcr, sai->base); + stm32_sai_pclk_disable(dev); + + return pinctrl_pm_select_default_state(dev); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_sai_suspend, stm32_sai_resume) +}; + MODULE_DEVICE_TABLE(of, stm32_sai_ids); static struct platform_driver stm32_sai_driver = { .driver = { .name = "st,stm32-sai", .of_match_table = stm32_sai_ids, + .pm = &stm32_sai_pm_ops, }, .probe = stm32_sai_probe, }; diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index 08de899c766b..9c36a393ab7b 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -268,6 +268,7 @@ struct stm32_sai_conf { * @version: SOC version * @irq: SAI interrupt line * @set_sync: pointer to synchro mode configuration callback + * @gcr: SAI Global Configuration Register */ struct stm32_sai_data { struct platform_device *pdev; @@ -279,4 +280,5 @@ struct stm32_sai_data { int irq; int (*set_sync)(struct stm32_sai_data *sai, struct device_node *np_provider, int synco, int synci); + u32 gcr; }; diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 55d802f51c15..2a90641cf164 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -168,6 +168,7 @@ static bool stm32_sai_sub_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case STM_SAI_DR_REGX: + case STM_SAI_SR_REGX: return true; default: return false; @@ -182,7 +183,6 @@ static bool stm32_sai_sub_writeable_reg(struct device *dev, unsigned int reg) case STM_SAI_FRCR_REGX: case STM_SAI_SLOTR_REGX: case STM_SAI_IMR_REGX: - case STM_SAI_SR_REGX: case STM_SAI_CLRFR_REGX: case STM_SAI_DR_REGX: case STM_SAI_PDMCR_REGX: @@ -202,6 +202,7 @@ static const struct regmap_config stm32_sai_sub_regmap_config_f4 = { .volatile_reg = stm32_sai_sub_volatile_reg, .writeable_reg = stm32_sai_sub_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { @@ -213,6 +214,7 @@ static const struct regmap_config stm32_sai_sub_regmap_config_h7 = { .volatile_reg = stm32_sai_sub_volatile_reg, .writeable_reg = stm32_sai_sub_writeable_reg, .fast_io = true, + .cache_type = REGCACHE_FLAT, }; static int snd_pcm_iec958_info(struct snd_kcontrol *kcontrol, @@ -441,8 +443,8 @@ static irqreturn_t stm32_sai_isr(int irq, void *devid) if (!flags) return IRQ_NONE; - regmap_update_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, - SAI_XCLRFR_MASK); + regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK, + SAI_XCLRFR_MASK); if (!sai->substream) { dev_err(&pdev->dev, "Device stopped. Spurious IRQ 0x%x\n", sr); @@ -704,9 +706,8 @@ static int stm32_sai_startup(struct snd_pcm_substream *substream, } /* Enable ITs */ - - regmap_update_bits(sai->regmap, STM_SAI_CLRFR_REGX, - SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); + regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, + SAI_XCLRFR_MASK, SAI_XCLRFR_MASK); imr = SAI_XIMR_OVRUDRIE; if (STM_SAI_IS_CAPTURE(sai)) { @@ -738,10 +739,10 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, * SAI fifo threshold is set to half fifo, to keep enough space * for DMA incoming bursts. */ - regmap_update_bits(sai->regmap, STM_SAI_CR2_REGX, - SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, - SAI_XCR2_FFLUSH | - SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); + regmap_write_bits(sai->regmap, STM_SAI_CR2_REGX, + SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK, + SAI_XCR2_FFLUSH | + SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF)); /* DS bits in CR1 not set for SPDIF (size forced to 24 bits).*/ if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) { @@ -1492,10 +1493,34 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int stm32_sai_sub_suspend(struct device *dev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + return 0; +} + +static int stm32_sai_sub_resume(struct device *dev) +{ + struct stm32_sai_sub_data *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops stm32_sai_sub_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(stm32_sai_sub_suspend, stm32_sai_sub_resume) +}; + static struct platform_driver stm32_sai_sub_driver = { .driver = { .name = "st,stm32-sai-sub", .of_match_table = stm32_sai_sub_ids, + .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, }; -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: sirf-audio: Remove redundant of_match_node call" to the asoc tree
by Mark Brown 25 Mar '19

25 Mar '19

The patch ASoC: sirf-audio: Remove redundant of_match_node call 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 78540a259b050480d20354efaf1055c233728b3b Mon Sep 17 00:00:00 2001 From: Aditya Pakki <pakki001(a)umn.edu> Date: Fri, 22 Mar 2019 17:00:29 -0500 Subject: [PATCH] ASoC: sirf-audio: Remove redundant of_match_node call Unlike other drivers probe method, of_match_node return value is not used or checked. This patch removes the redundant code. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Steven Price <steven.price(a)arm.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sirf-audio-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index e424499a8450..e0af21050078 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -461,9 +461,6 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) struct sirf_audio_codec *sirf_audio_codec; void __iomem *base; struct resource *mem_res; - const struct of_device_id *match; - - match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix NULL pointer dereference in snd_soc_dapm_free_kcontrol
by Pankaj Bharadiya 25 Mar '19

25 Mar '19

w_text_param can be NULL and it is being dereferenced without checking. Add the missing sanity check to prevent NULL pointer dereference. Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya(a)intel.com> --- sound/soc/soc-dapm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ec06ef..67b032c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3957,6 +3957,10 @@ snd_soc_dapm_free_kcontrol(struct snd_soc_card *card, int count; devm_kfree(card->dev, (void *)*private_value); + + if (!w_param_text) + return; + for (count = 0 ; count < num_params; count++) devm_kfree(card->dev, (void *)w_param_text[count]); devm_kfree(card->dev, w_param_text); -- 2.7.4

3 2

[alsa-devel] [v2] ASoC: AMD: Configure wclk and bclk of master codec
by Agrawal, Akshu 25 Mar '19

25 Mar '19

With CCF support in da7219, we can now set the correct rate of wclk and bclk. Setting bclk at lower rate at 1.53Mhz from its earlier default rate of 3Mhz, also fixes noise issue observed on some dmics. v2: Removed clk set rate and enable/disable from da7219 ops as da7219 codec takes care of them internally. Clock configuration kept for those codecs where da7219 acts as master of clock. As suggested by Adam Thomson. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 45 +++++++++++++++++----------- 1 file changed, 28 insertions(+), 17 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8e2ce369a295..7737c6dc026a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -46,7 +46,8 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -static struct clk *da7219_dai_clk; +static struct clk *da7219_dai_wclk; +static struct clk *da7219_dai_bclk; extern int bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -72,7 +73,8 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -94,12 +96,15 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream) +static int da7219_clk_enable(struct snd_pcm_substream *substream, + int wclk_rate, int bclk_rate) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - ret = clk_prepare_enable(da7219_dai_clk); + clk_set_rate(da7219_dai_wclk, wclk_rate); + clk_set_rate(da7219_dai_bclk, bclk_rate); + ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); return ret; @@ -110,7 +115,7 @@ static int da7219_clk_enable(struct snd_pcm_substream *substream) static void da7219_clk_disable(void) { - clk_disable_unprepare(da7219_dai_clk); + clk_disable_unprepare(da7219_dai_bclk); } static const unsigned int channels[] = { @@ -151,7 +156,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -173,12 +178,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return da7219_clk_enable(substream); -} - -static void cz_da7219_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,7 +199,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static void cz_max_shutdown(struct snd_pcm_substream *substream) @@ -225,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -247,7 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + return 0; } static void cz_dmic_shutdown(struct snd_pcm_substream *substream) @@ -255,29 +255,40 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int wclk, bclk; + + wclk = params_rate(params); + bclk = wclk * params_channels(params) * + snd_pcm_format_width(params_format(params)); + return da7219_clk_enable(substream, wclk, bclk); +} 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_cap_startup, - .shutdown = cz_da7219_shutdown, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

2 2

[alsa-devel] [PATCH v3 00/11] ASoC: tlv320aic32x4: Rework Clock Setting
by Annaliese McDermond 25 Mar '19

25 Mar '19

The current tlv320aic32x4 code sets the various clock parameters by looking in a static table in the kernel. This works well enough, but it has a few disadvantages. 1) If the master clock doesn't match one of the precalculated values the driver cannot work. This could be because the designer decided to use a different crystal, or because the clock comes from a PLL that can't quite achieve the proper frequency exactly. 2) The driver only supports certain pre-calculated sample rates. The actual hardware can support many more. These changes enable those rates. Additionally, certain sample rates at certain clock rates were previously unavailable as combinations. This patch dynamically calculates the various PLL and divider values to find something that works. Additionally, to enable much of this, the clock tree is modeled as a part of the Common Clock Framework. This allows for easier debugigng as you can use the standard tools such as /sys/kernel/debug/clk/clk_summary. It also simplifies enabling the entirety of the clock tree needed, and makes sure unused clocks become disabled. These patches have been tested and are working on a Raspberry Pi with the simplecard driver. Changes in v2: - Update 0002 patch to change Kconfig to add dependency on CCF - Update 0002 patch to correct the SPDX identifier - Update 0002 patch to use EXPORT_SYMBOL_GPL() instead of EXPORT_SYMBOL() - Update patches 0002, 0004 and 0005 to use prepare/unprepare instead of enable/disable since the i2c operations are not atomic - Update 0004 patch to fix indentation in commit message - Drop patch 0009 because it is a duplicate of the already committed patch in 667e9334 - Update 0019 patch to keep aic32x4_set_dai_sysclk but to have it use clk_set_rate to set the master clock rate Changes in v3: - Rebase on current for-next - Move "Properly Set Processing Blocks" to the start of the patches so that it can potentially be applied as a bug fix Annaliese McDermond (11): ASoC: tlv320aic32x4: Properly Set Processing Blocks ASoC: tlv320aic32x4: Model PLL in CCF ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF ASoC: tlv320aic32x4: Model DAC/ADC dividers in CCF ASoC: tlv320aic32x4: Model BDIV divider in CCF ASoC: tlv320aic32x4: Control clock gating with CCF ASoC: tlv320aic32x4: Move aosr and dosr setting to separate functions ASoC: tlv320aic32x4: Dynamically Determine Clocking ASoC: tlv320aic32x4: Restructure set_dai_sysclk ASoC: tlv320aic32x4: Remove mclk references ASoC: tlv320aic32x4: Allow 192000 Sample Rate sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tlv320aic32x4-clk.c | 483 +++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 357 ++++++++++---------- sound/soc/codecs/tlv320aic32x4.h | 11 + 5 files changed, 665 insertions(+), 189 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4-clk.c -- 2.19.1

2 15

[alsa-devel] [PATCH 1/3] ASoC: ti: remove compat dma probing
by Arnd Bergmann 25 Mar '19

25 Mar '19

After running into a link error: sound/soc/ti/edma-pcm.o:(.rodata+0x18): undefined reference to `edma_filter_fn' I checked all users of this, and they have new-style 'dma_slave_map' tables, so none of them should still need it. Removing the associated lines simplifies the code and avoids the build-time dependency on the respective dmaengine drivers. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: remove header inclusions as well I added the two dmaengine patches to the series this time, but we probably want to have those merged separately through the dmaengine tree, while this one should get merged as a bugfix through ASoC. --- sound/soc/ti/edma-pcm.c | 5 +---- sound/soc/ti/sdma-pcm.c | 9 +++------ 2 files changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/ti/edma-pcm.c b/sound/soc/ti/edma-pcm.c index 59e588abe54b..fdffb801b185 100644 --- a/sound/soc/ti/edma-pcm.c +++ b/sound/soc/ti/edma-pcm.c @@ -23,7 +23,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <linux/edma.h> #include "edma-pcm.h" @@ -43,14 +42,12 @@ static const struct snd_pcm_hardware edma_pcm_hardware = { static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = { .pcm_hardware = &edma_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = edma_filter_fn, .prealloc_buffer_size = 128 * 1024, }; int edma_pcm_platform_register(struct device *dev) { - return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, 0); } EXPORT_SYMBOL_GPL(edma_pcm_platform_register); diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index 21a9c2499d48..a236350beb10 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -11,7 +11,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <linux/omap-dmaengine.h> #include "sdma-pcm.h" @@ -31,7 +30,6 @@ static const struct snd_pcm_hardware sdma_pcm_hardware = { static const struct snd_dmaengine_pcm_config sdma_dmaengine_pcm_config = { .pcm_hardware = &sdma_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = omap_dma_filter_fn, .prealloc_buffer_size = 128 * 1024, }; @@ -39,13 +37,12 @@ int sdma_pcm_platform_register(struct device *dev, char *txdmachan, char *rxdmachan) { struct snd_dmaengine_pcm_config *config; - unsigned int flags = SND_DMAENGINE_PCM_FLAG_COMPAT; + unsigned int flags = 0; /* Standard names for the directions: 'tx' and 'rx' */ if (!txdmachan && !rxdmachan) return devm_snd_dmaengine_pcm_register(dev, - &sdma_dmaengine_pcm_config, - flags); + &sdma_dmaengine_pcm_config, 0); config = devm_kzalloc(dev, sizeof(*config), GFP_KERNEL); if (!config) @@ -65,7 +62,7 @@ int sdma_pcm_platform_register(struct device *dev, config->chan_names[0] = txdmachan; config->chan_names[1] = rxdmachan; - return devm_snd_dmaengine_pcm_register(dev, config, flags); + return devm_snd_dmaengine_pcm_register(dev, config, 0); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); -- 2.20.0

4 9

[alsa-devel] [PATCH] ALSA: hda - add Lenovo IdeaCentre B550 to the power_save_blacklist
by Jaroslav Kysela 25 Mar '19

25 Mar '19

Another machine which does not like the power saving (noise): https://bugzilla.redhat.com/show_bug.cgi?id=1689623 Also, reorder the Lenovo C50 entry to keep the table sorted. Reported-by: hs.guimaraes(a)outlook.com Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e5c49003e75f..4f502c92061f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2144,10 +2144,12 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), - /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ - SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1689623 */ + SND_PCI_QUIRK(0x17aa, 0x367b, "Lenovo IdeaCentre B550", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ + SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), {} }; #endif /* CONFIG_PM */ -- 2.20.1

3 2

[alsa-devel] [PATCH] sound: codecs: fix a missing check for regmap_update_bits
by Kangjie Lu 25 Mar '19

25 Mar '19

regmap_update_bits could fail. The fix checks its status and if it fails, returns its error code upstream. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> --- sound/soc/codecs/cs35l34.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 5063c05afa27..a456b1ac9019 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -262,6 +262,8 @@ static int cs35l34_sdin_event(struct snd_soc_dapm_widget *w, } ret = regmap_update_bits(priv->regmap, CS35L34_PWRCTL1, CS35L34_PDN_ALL, CS35L34_PDN_ALL); + if (ret < 0) + return ret; break; default: pr_err("Invalid event = 0x%x\n", event); -- 2.17.1

1 0

[alsa-devel] Noise on alc285
by José Ramón Muñoz Pekkarinen 24 Mar '19

24 Mar '19

Hi, Can I have some help fixing noise problems in my hp envy x360(ag0808no)?. The description of the machine is uploaded at [1]. I tried both from my gentoo system as specified in the link, as well as using my dual booted ubuntu bionic, both reproduce noise doing a short record, as well as changing the headphones for several different sets. Changing position-fix parameter in snd-hda-intel or model to lenovo-spk-noise or dell-spk-noise didn't mitigate the issue. I tried also alc295-hp-x360 and alc280-hp-headset with same luck. [1]http://alsa-project.org/db/?f=22fd613b2577a3d3dc99a31f5cf04420bc688615 Thanks! José.

1 0

Re: [alsa-devel] KASAN: use-after-free Read in rt_cache_valid
by syzbot 23 Mar '19

23 Mar '19

syzbot has bisected this bug to: commit 6b27e27729270a2478fdebea2db9c4f57bb4e8e6 Author: Colin Ian King <colin.king(a)canonical.com> Date: Wed Oct 31 19:31:43 2018 +0000 ASoC: stm32: sai: fix less than zero comparison on unsigned int bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=10c0298b200000 start commit: aa0c38cf Merge branch 'fixes' of git://git.kernel.org/pub/.. git tree: upstream final crash: https://syzkaller.appspot.com/x/report.txt?x=12c0298b200000 console output: https://syzkaller.appspot.com/x/log.txt?x=14c0298b200000 kernel config: https://syzkaller.appspot.com/x/.config?x=ee434566c893c7b1 dashboard link: https://syzkaller.appspot.com/bug?extid=c4c4b2bb358bb936ad7e syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1197cb88c00000 Reported-by: syzbot+c4c4b2bb358bb936ad7e(a)syzkaller.appspotmail.com Fixes: 6b27e2772927 ("ASoC: stm32: sai: fix less than zero comparison on unsigned int") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: enable S24_LE format support
by Jenny TC 23 Mar '19

23 Mar '19

From: Jenny TC <jenny.tc(a)intel.com> To enable S24_LE format, sample_type in topology fw has to be set to 1. But sample_type defined in topology firmware configuration is not getting reflected in the dsp param. This patch sets sample_type in base config so that the sample type defined in the topology firmware is reflected in the dsp params. This issues was uncovered while debugging the S24_LE format which require the MSB byte in 32 bit word to be skipped. Setting sample_type in topology firmware to 1 helps to skip MSB byte word. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 28c4806..4bf70b4 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -483,6 +483,7 @@ static void skl_set_base_module_format(struct skl_sst *ctx, base_cfg->audio_fmt.bit_depth = format->bit_depth; base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; base_cfg->audio_fmt.ch_cfg = format->ch_cfg; + base_cfg->audio_fmt.sample_type = format->sample_type; dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", format->bit_depth, format->valid_bit_depth, -- 1.9.1

2 1

Re: [alsa-devel] [ALSA patch] [PATCH] [-, JACK, plugin, 1/1] configure: use $prefix/etc instead of /etc
by Takashi Iwai 23 Mar '19

23 Mar '19

On Wed, 20 Mar 2019 18:30:47 +0100, Alex Ivanov wrote: > > Use $sysconfdir as prefix for ALSA_LCONF_DIR by default. Otherwise install > fails on non-FHS distros. > > Signed-off-by: Alex Ivanov <gnidorah(a)ya.ru> > --- > configure.ac | 19 ++++++++++++++++++- > 1 file changed, 18 insertions(+), 1 deletion(-) > > diff --git a/configure.ac b/configure.ac > index 5b80585..7b8dd66 100644 > --- a/configure.ac > +++ b/configure.ac > @@ -219,6 +219,23 @@ AC_DEFINE_UNQUOTED(ALSA_DATA_DIR, "$alsadatadir", [directory containing ALSA dat > ALSA_DATA_DIR="$alsadatadir" > AC_SUBST(ALSA_DATA_DIR) > > +dnl ALSA configuration directory > +AC_ARG_WITH(alsaconfdir, > + AS_HELP_STRING([--with-alsaconfdir=dir], > + [path where ALSA configuration files are stored]), > + alsaconfdir="$withval", alsaconfdir="") > +if test -z "$alsaconfdir"; then > + eval dir="$sysconfdir" > + case "$dir" in > + /*) ;; > + *) dir="$dir" > + esac > + alsaconfdir="$dir/alsa" > +fi > +AC_DEFINE_UNQUOTED(ALSA_CONF_DIR, "$alsaconfdir", [directory containing ALSA configuration files]) > +ALSA_CONF_DIR="$alsaconfdir" > +AC_SUBST(ALSA_CONF_DIR) Yet another option is way too confusing, IMO. > + > dnl ALSA add-on global config directory > AC_ARG_WITH(alsagconfdir, > AS_HELP_STRING([--with-alsagconfdir=dir], > @@ -237,7 +254,7 @@ AC_ARG_WITH(alsalconfdir, > [path where ALSA local add-on config files are stored]), > alsalconfdir="$withval", alsalconfdir="") > if test -z "$alsalconfdir"; then > - alsalconfdir="/etc/alsa/conf.d" > + alsalconfdir="$ALSA_CONF_DIR/conf.d" Let's add the $sysconfdir usage here instead of adding a new --with-*. Care to resubmit with that change? thanks, Takashi

2 3

[alsa-devel] Hercules P32 DJ: severe sound distortion during MIDI transfers
by Guido Trentalancia 22 Mar '19

22 Mar '19

Hello. I am hitting a very serious bug (ALSA kernel driver or ALSA library) when using the Hercules P32 DJ audio interface. The sound is severely distorted during MIDI transfers. To reproduce: - start playing something in a first console: console1# AUDIODEV=hw:2,0 play audio.wav - the audio plays fine - now start "amidi" in a second console while the above track is still playing console2# amidi -p hw:2,0,0 -d - the sound is now severely distorted (basically noise, with some hard- to-distinguish features resembling the original track) until "amidi" is interrupted ! Kernel versions tested: from 4.12 up to 5.0.2 (latest). NOTES: Always reproducible. Same device is working fine on Windows. The output from alsa-info.sh and lsusb -v follows. upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Wed Mar 20 20:44:36 UTC 2019 !!Linux Distribution !!------------------ !!DMI Information !!--------------- Manufacturer: Acer Product Name: Aspire 5745G Product Version: V1.19 Firmware Version: V1.19 Board Vendor: Acer Board Name: JV51_CP !!ACPI Device Status Information !!--------------- /sys/bus/acpi/devices/PNP0103:00/status 15 /sys/bus/acpi/devices/PNP0C0A:00/status 31 /sys/bus/acpi/devices/PNP0C0F:00/status 9 /sys/bus/acpi/devices/PNP0C0F:01/status 9 /sys/bus/acpi/devices/PNP0C0F:02/status 9 /sys/bus/acpi/devices/PNP0C0F:03/status 9 /sys/bus/acpi/devices/PNP0C0F:04/status 9 /sys/bus/acpi/devices/PNP0C0F:05/status 9 /sys/bus/acpi/devices/PNP0C0F:06/status 9 /sys/bus/acpi/devices/PNP0C0F:07/status 9 !!Kernel Information !!------------------ Kernel release: 5.0.2 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k5.0.2 Library version: 1.1.8 Utilities version: 1.1.8 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel snd_usb_audio !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [MID ]: HDA-Intel - HDA Intel MID HDA Intel MID at 0xb7100000 irq 27 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xb3000000 irq 16 2 [DJ ]: USB-Audio - Hercules P32 DJ Guillemot Corporation Hercules P32 DJ at usb- 0000:00:1d.0-1.4, full speed !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) 01:00.1 Audio device: NVIDIA Corporation GT216 HDMI Audio Controller (rev a1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:3b56 (rev 05) Subsystem: 1025:035b -- 01:00.1 0403: 10de:0be2 (rev a1) Subsystem: 1025:035a !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) pm_blacklist : Y position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : -1 snoop : -1 !!Module: snd_usb_audio autoclock : Y device_setup : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) ignore_ctl_error : N index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 pid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 quirk_alias : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),( null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(n ull),(null),(null),(null),(null),(null),(null),(null),(null),(null),(nu ll),(null) use_vmalloc : Y vid : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,- 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC269VB Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0269 Subsystem Id: 0x1025035b Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC269VB Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x51 0x51] Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC269VB Digital", type="SPDIF", device=1 Converter: stream=5, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x8b 0x8b] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC269VB Analog", type="Audio", device=0 Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=4, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x18 0x19 0x1a 0x1b 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x03 0x0b Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] [0x80] Connection: 2 0x02 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x40000b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x99a30930: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010014: OUT EAPD Detect EAPD 0x2: EAPD Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 2 0x0c 0x0d* Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03a19840: [Jack] Mic at Ext Left Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x0d Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001724: IN Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0d Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4017992d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x03451120: [Jack] SPDIF Out at Ext Left Conn = Optical, Color = Black DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=25 Node 0x21 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Connection: 2 0x0c* 0x0d Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 7 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12* Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x18 0x19 0x1a 0x1b 0x1d 0x0b Codec: Nvidia Generic Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de000a Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Device: name="Generic Digital", type="HDMI", device=3 Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x04 Codec: Nvidia Generic Address: 1 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de000a Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Control: name="IEC958 Playback Con Mask", index=1, device=0 Control: name="IEC958 Playback Pro Mask", index=1, device=0 Control: name="IEC958 Playback Default", index=1, device=0 Control: name="IEC958 Playback Switch", index=1, device=0 Device: name="Generic Digital", type="HDMI", device=7 Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x04 Codec: Nvidia Generic Address: 2 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de000a Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Control: name="IEC958 Playback Con Mask", index=2, device=0 Control: name="IEC958 Playback Pro Mask", index=2, device=0 Control: name="IEC958 Playback Default", index=2, device=0 Control: name="IEC958 Playback Switch", index=2, device=0 Device: name="Generic Digital", type="HDMI", device=8 Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x04 Codec: Nvidia Generic Address: 3 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10de000a Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Control: name="IEC958 Playback Con Mask", index=3, device=0 Control: name="IEC958 Playback Pro Mask", index=3, device=0 Control: name="IEC958 Playback Default", index=3, device=0 Control: name="IEC958 Playback Switch", index=3, device=0 Device: name="Generic Digital", type="HDMI", device=9 Converter: stream=5, channel=0 Digital: Enabled Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HBR HDMI DP Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Connection: 1 0x04 --endcollapse-- !!USB Mixer information !!--------------------- --startcollapse-- USB Mixer: usb_id=0x06f8b121, ctrlif=0, ctlerr=0 Card: Guillemot Corporation Hercules P32 DJ at usb-0000:00:1d.0-1.4, full speed Unit: 2 Control: name="PCM Playback Volume", index=0 Info: id=2, control=2, cmask=0xf, channels=4, type="S16" Volume: min=-25600, max=0, dBmin=-10000, dBmax=0 Unit: 2 Control: name="PCM Playback Switch", index=0 Info: id=2, control=1, cmask=0xf, channels=4, type="INV_BOOLEAN" Volume: min=0, max=1, dBmin=0, dBmax=0 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----. 1 root audio 116, 6 Mar 20 18:15 /dev/snd/controlC0 crw-rw----. 1 root audio 116, 15 Mar 20 18:15 /dev/snd/controlC1 crw-rw----. 1 root audio 116, 18 Mar 20 21:25 /dev/snd/controlC2 crw-rw----. 1 root audio 116, 5 Mar 20 18:15 /dev/snd/hwC0D0 crw-rw----. 1 root audio 116, 11 Mar 20 18:15 /dev/snd/hwC1D0 crw-rw----. 1 root audio 116, 12 Mar 20 18:15 /dev/snd/hwC1D1 crw-rw----. 1 root audio 116, 13 Mar 20 18:15 /dev/snd/hwC1D2 crw-rw----. 1 root audio 116, 14 Mar 20 18:15 /dev/snd/hwC1D3 crw-rw----. 1 root audio 116, 17 Mar 20 21:25 /dev/snd/midiC2D0 crw-rw----. 1 root audio 116, 3 Mar 20 19:58 /dev/snd/pcmC0D0c crw-rw----. 1 root audio 116, 2 Mar 20 20:40 /dev/snd/pcmC0D0p crw-rw----. 1 root audio 116, 4 Mar 20 19:58 /dev/snd/pcmC0D1p crw-rw----. 1 root audio 116, 7 Mar 20 19:58 /dev/snd/pcmC1D3p crw-rw----. 1 root audio 116, 8 Mar 20 19:58 /dev/snd/pcmC1D7p crw-rw----. 1 root audio 116, 9 Mar 20 19:58 /dev/snd/pcmC1D8p crw-rw----. 1 root audio 116, 10 Mar 20 19:58 /dev/snd/pcmC1D9p crw-rw----. 1 root audio 116, 16 Mar 20 21:32 /dev/snd/pcmC2D0p crw-rw----. 1 root audio 116, 1 Mar 20 17:19 /dev/snd/seq crw-rw----. 1 root audio 116, 33 Mar 20 18:15 /dev/snd/timer /dev/snd/by-id: total 0 drwxr-xr-x. 2 root root 60 Mar 20 21:25 . drwxr-xr-x. 4 root root 460 Mar 20 21:25 .. lrwxrwxrwx. 1 root root 12 Mar 20 21:25 usb- Guillemot_Corporation_Hercules_P32_DJ-00 -> ../controlC2 /dev/snd/by-path: total 0 drwxr-xr-x. 2 root root 100 Mar 20 21:25 . drwxr-xr-x. 4 root root 460 Mar 20 21:25 .. lrwxrwxrwx. 1 root root 12 Mar 20 18:15 pci-0000:00:1b.0 -> ../controlC0 lrwxrwxrwx. 1 root root 12 Mar 20 21:25 pci-0000:00:1d.0-usb-0:1.4:1.0 -> ../controlC2 lrwxrwxrwx. 1 root root 12 Mar 20 18:15 pci-0000:01:00.1 -> ../controlC1 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: MID [HDA Intel MID], device 1: ALC269VB Digital [ALC269VB Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: DJ [Hercules P32 DJ], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: MID [HDA Intel MID], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [MID] Card hw:0 'MID'/'HDA Intel MID at 0xb7100000 irq 27' Mixer name : 'Realtek ALC269VB' Components : 'HDA:10ec0269,1025035b,00100100' Controls : 31 Simple ctrls : 14 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 81 [93%] [-4.50dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback 0 [0%] [-65.25dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 128 [50%] [-25.40dB] Front Right: Playback 128 [50%] [-25.40dB] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-16.50dB] [on] Front Right: Capture 0 [0%] [-16.50dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' !!-------Mixer controls for card 1 [NVidia] Card hw:1 'NVidia'/'HDA NVidia at 0xb3000000 irq 16' Mixer name : 'Nvidia Generic' Components : 'HDA:10de000a,10de0101,00100100' Controls : 24 Simple ctrls : 4 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',2 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958',3 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card 2 [DJ] Card hw:2 'DJ'/'Guillemot Corporation Hercules P32 DJ at usb- 0000:00:1d.0-1.4, full speed' Mixer name : 'USB Mixer' Components : 'USB06f8:b121' Controls : 4 Simple ctrls : 1 Simple mixer control 'PCM',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right - Rear Left - Rear Right Limits: Playback 0 - 400 Mono: Front Left: Playback 320 [80%] [-20.00dB] [on] Front Right: Playback 320 [80%] [-20.00dB] [on] Rear Left: Playback 320 [80%] [-20.00dB] [on] Rear Right: Playback 320 [80%] [-20.00dB] [on] !!Alsactl output !!-------------- --startcollapse-- state.MID { control.1 { iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -6525 dbvalue.1 -6525 } } control.2 { iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Speaker Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Loopback Mixing' value Disabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.6 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.7 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Auto-Mute Mode' value Enabled comment { access 'read write' type ENUMERATED count 1 item.0 Disabled item.1 Enabled } } control.9 { iface MIXER name 'Capture Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1650 dbmax 3000 dbvalue.0 -1650 dbvalue.1 -1650 } } control.10 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.11 { iface MIXER name 'Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3600 dbvalue.0 0 dbvalue.1 0 } } control.12 { iface MIXER name 'Internal Mic Boost Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 3' dbmin 0 dbmax 3600 dbvalue.0 0 dbvalue.1 0 } } control.13 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.14 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.17 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.18 { iface MIXER name 'IEC958 Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.19 { iface MIXER name 'IEC958 Default PCM Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.20 { iface MIXER name 'Master Playback Volume' value 81 comment { access 'read write' type INTEGER count 1 range '0 - 87' dbmin -6525 dbmax 0 dbvalue.0 -450 } } control.21 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.22 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } control.23 { iface CARD name 'Internal Mic Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.24 { iface CARD name 'Headphone Jack' value false comment { access read type BOOLEAN count 1 } } control.25 { iface CARD name 'Speaker Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.26 { iface CARD name 'SPDIF Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.27 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.28 { iface PCM name 'Capture Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.29 { iface PCM device 1 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.30 { iface MIXER name 'PCM Playback Volume' value.0 128 value.1 128 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -2540 dbvalue.1 -2540 } } control.31 { iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 0 dbvalue.1 0 } } } state.NVidia { control.1 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.2 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.3 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.4 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.5 { iface CARD name 'HDMI Jack' value false comment { access read type BOOLEAN count 1 } } control.6 { iface PCM device 3 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.7 { iface MIXER name 'IEC958 Playback Con Mask' index 1 value '0fff000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.8 { iface MIXER name 'IEC958 Playback Pro Mask' index 1 value '0f00000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.9 { iface MIXER name 'IEC958 Playback Default' index 1 value '0400000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.10 { iface MIXER name 'IEC958 Playback Switch' index 1 value true comment { access 'read write' type BOOLEAN count 1 } } control.11 { iface CARD name 'HDMI Jack' index 1 value false comment { access read type BOOLEAN count 1 } } control.12 { iface PCM device 7 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.13 { iface MIXER name 'IEC958 Playback Con Mask' index 2 value '0fff000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.14 { iface MIXER name 'IEC958 Playback Pro Mask' index 2 value '0f00000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.15 { iface MIXER name 'IEC958 Playback Default' index 2 value '0400000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.16 { iface MIXER name 'IEC958 Playback Switch' index 2 value true comment { access 'read write' type BOOLEAN count 1 } } control.17 { iface CARD name 'HDMI Jack' index 2 value false comment { access read type BOOLEAN count 1 } } control.18 { iface PCM device 8 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } control.19 { iface MIXER name 'IEC958 Playback Con Mask' index 3 value '0fff000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.20 { iface MIXER name 'IEC958 Playback Pro Mask' index 3 value '0f00000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.21 { iface MIXER name 'IEC958 Playback Default' index 3 value '0400000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 00000000000000000000000000000000000000000000000000000000000000000000000 000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.22 { iface MIXER name 'IEC958 Playback Switch' index 3 value true comment { access 'read write' type BOOLEAN count 1 } } control.23 { iface CARD name 'HDMI Jack' index 3 value false comment { access read type BOOLEAN count 1 } } control.24 { iface PCM device 9 name 'Playback Channel Map' value.0 0 value.1 0 comment { access read type INTEGER count 2 range '0 - 36' } } } state.DJ { control.1 { iface PCM name 'Playback Channel Map' value.0 0 value.1 0 value.2 0 value.3 0 comment { access read type INTEGER count 4 range '0 - 36' } } control.2 { iface MIXER name 'PCM Playback Switch' value.0 true value.1 true value.2 true value.3 true comment { access 'read write' type BOOLEAN count 4 } } control.3 { iface MIXER name 'PCM Playback Volume' value.0 320 value.1 320 value.2 320 value.3 320 comment { access 'read write' type INTEGER count 4 range '0 - 400' dbmin -10000 dbmax 0 dbvalue.0 -2000 dbvalue.1 -2000 dbvalue.2 -2000 dbvalue.3 -2000 } } control.4 { iface CARD name 'Keep Interface' value false comment { access 'read write' type BOOLEAN count 1 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_hrtimer snd_seq_midi snd_seq_dummy snd_seq_midi_event snd_seq snd_pcm_oss snd_mixer_oss xt_iprange ipv6 crc_ccitt arc4 brcmsmac cordic brcmutil mac80211 sha256_ssse3 snd_usb_audio snd_usbmidi_lib sha256_generic snd_rawmidi cfg80211 snd_seq_device iTCO_wdt iTCO_vendor_support snd_hda_codec_realtek snd_hda_codec_generic ledtrig_audio snd_hda_intel joydev snd_hda_codec acer_wmi sparse_keymap bcma snd_hda_core rfkill pcc_cpufreq snd_hwdep acpi_cpufreq snd_pcm snd_timer mei_me snd mei pcspkr soundcore lpc_ich i2c_i801 mfd_core ums_onetouch usb_storage dm_crypt serio_raw atl1c nouveau mxm_wmi i2c_algo_bit drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops ttm wmi video drm i2c_core !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x12 0x99a30930 0x14 0x99130110 0x17 0x411111f0 0x18 0x03a19840 0x19 0x411111f0 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x4017992d 0x1e 0x03451120 0x21 0x0321101f /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC1D0/init_pin_configs: 0x05 0x18560010 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: /sys/class/sound/hwC1D0/hints: /sys/class/sound/hwC1D1/init_pin_configs: 0x05 0x18560010 /sys/class/sound/hwC1D1/driver_pin_configs: /sys/class/sound/hwC1D1/user_pin_configs: /sys/class/sound/hwC1D1/init_verbs: /sys/class/sound/hwC1D1/hints: /sys/class/sound/hwC1D2/init_pin_configs: 0x05 0x18560010 /sys/class/sound/hwC1D2/driver_pin_configs: /sys/class/sound/hwC1D2/user_pin_configs: /sys/class/sound/hwC1D2/init_verbs: /sys/class/sound/hwC1D2/hints: /sys/class/sound/hwC1D3/init_pin_configs: 0x05 0x18560010 /sys/class/sound/hwC1D3/driver_pin_configs: /sys/class/sound/hwC1D3/user_pin_configs: /sys/class/sound/hwC1D3/init_verbs: /sys/class/sound/hwC1D3/hints: !!ALSA/HDA dmesg !!-------------- [ 0.238633] ACPI: Added _OSI(Linux-Dell-Video) [ 0.238779] ACPI: Added _OSI(Linux-Lenovo-NV-HDMI-Audio) [ 0.238926] ACPI: Added _OSI(Linux-HPI-Hybrid-Graphics) -- [ 24.240651] input: Acer WMI hotkeys as /devices/virtual/input/input16 [ 24.294387] snd_hda_intel 0000:00:1b.0: enabling device (0000 -> 0002) [ 24.295148] snd_hda_intel 0000:01:00.1: Disabling MSI [ 24.417727] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC269VB: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 24.418128] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.418507] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 24.419057] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 24.419695] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 24.420312] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 24.420959] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 24.421602] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 [ 24.448267] input: HDA Intel MID Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input17 [ 24.448762] input: HDA Intel MID Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input18 [ 24.517380] mei mei::55213584-9a29-4916-badf-0fb7ed682aeb:01: Could not read FW version [ 24.517783] mei mei::55213584-9a29-4916-badf-0fb7ed682aeb:01: FW version command failed -5 [ 24.606385] snd_hda_codec_generic hdaudioC1D0: autoconfig for Generic: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 24.606759] snd_hda_codec_generic hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.607115] snd_hda_codec_generic hdaudioC1D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.607684] snd_hda_codec_generic hdaudioC1D0: mono: mono_out=0x0 [ 24.608266] snd_hda_codec_generic hdaudioC1D0: dig-out=0x5/0x0 [ 24.608871] snd_hda_codec_generic hdaudioC1D0: inputs: [ 24.684106] mce: CPU3: Core temperature above threshold, cpu clock throttled (total events = 1) -- [ 24.770560] iTCO_vendor_support: vendor-support=0 [ 24.790526] snd_hda_codec_generic hdaudioC1D1: autoconfig for Generic: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 24.790874] snd_hda_codec_generic hdaudioC1D1: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.791202] snd_hda_codec_generic hdaudioC1D1: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.791536] snd_hda_codec_generic hdaudioC1D1: mono: mono_out=0x0 [ 24.791995] snd_hda_codec_generic hdaudioC1D1: dig-out=0x5/0x0 [ 24.792477] snd_hda_codec_generic hdaudioC1D1: inputs: [ 24.799553] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 -- [ 24.800621] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 24.850369] snd_hda_codec_generic hdaudioC1D2: autoconfig for Generic: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 24.850650] snd_hda_codec_generic hdaudioC1D2: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.850911] snd_hda_codec_generic hdaudioC1D2: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 24.851165] snd_hda_codec_generic hdaudioC1D2: mono: mono_out=0x0 [ 24.851415] snd_hda_codec_generic hdaudioC1D2: dig-out=0x5/0x0 [ 24.851747] snd_hda_codec_generic hdaudioC1D2: inputs: [ 25.056813] snd_hda_codec_generic hdaudioC1D3: autoconfig for Generic: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 25.057109] snd_hda_codec_generic hdaudioC1D3: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 25.057393] snd_hda_codec_generic hdaudioC1D3: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 25.057658] snd_hda_codec_generic hdaudioC1D3: mono: mono_out=0x0 [ 25.057924] snd_hda_codec_generic hdaudioC1D3: dig-out=0x5/0x0 [ 25.058180] snd_hda_codec_generic hdaudioC1D3: inputs: [ 25.094153] input: HDA NVidia HDMI as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input19 [ 25.094539] input: HDA NVidia HDMI as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input20 [ 25.094891] input: HDA NVidia HDMI as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input21 [ 25.095249] input: HDA NVidia HDMI as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input22 [ 25.783253] cfg80211: Loading compiled-in X.509 certificates for regulatory database -- [ 26.044956] usb 2-1.4: [2] FU [PCM Playback Volume] ch = 4, val = -25600/0/64 [ 26.046216] usbcore: registered new interface driver snd-usb-audio [ 26.088396] cfg80211: Problem loading in-kernel X.509 certificate (- 13) ----------------------- lsusb -v Bus 002 Device 003: ID 06f8:b121 Guillemot Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x06f8 Guillemot Corp. idProduct 0xb121 bcdDevice 2.00 iManufacturer 1 Guillemot Corporation iProduct 2 Hercules P32 DJ iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 196 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 480mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 43 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 4 wChannelConfig 0x0033 Left Front (L) Right Front (R) Left Surround (LS) Right Surround (RS) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 1 bControlSize 1 bmaControls( 0) 0x00 bmaControls( 1) 0x03 Mute Control Volume Control bmaControls( 2) 0x03 Mute Control Volume Control bmaControls( 3) 0x03 Mute Control Volume Control bmaControls( 4) 0x03 Mute Control Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 2 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 4 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0188 1x 392 bytes bInterval 1 bRefresh 0 bSynchAddress 129 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 17 Transfer Type Isochronous Synch Type None Usage Type Feedback wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 2 Hercules P32 DJ MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 65 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Device Status: 0x0e01 Self Powered Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x8087 Intel Corp. idProduct 0x0020 Integrated Rate Matching Hub bcdDevice 0.00 iManufacturer 0 iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 12 Hub Descriptor: bLength 11 bDescriptorType 41 nNbrPorts 8 wHubCharacteristic 0x0089 Per-port power switching Per-port overcurrent protection TT think time 8 FS bits Port indicators bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 0x00 PortPwrCtrlMask 0xff 0xff Hub Port Status: Port 1: 0000.0100 power Port 2: 0000.0100 power Port 3: 0000.0100 power Port 4: 0000.0103 power enable connect Port 5: 0000.0100 power Port 6: 0000.0100 power Port 7: 0000.0100 power Port 8: 0000.0100 power Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 5.00 iManufacturer 3 Linux 5.0.2 ehci_hcd iProduct 2 EHCI Host Controller iSerial 1 0000:00:1d.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 3 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0xff Hub Port Status: Port 1: 0000.0503 highspeed power enable connect Port 2: 0000.0100 power Port 3: 0000.0100 power Device Status: 0x0001 Self Powered Bus 001 Device 005: ID 0d49:7010 Maxtor OneTouch Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x0d49 Maxtor idProduct 0x7010 OneTouch bcdDevice 2.00 iManufacturer 1 Maxtor iProduct 3 OneTouch iSerial 2 Y61LWLCE bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 39 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 3 bInterfaceClass 8 Mass Storage bInterfaceSubClass 6 SCSI bInterfaceProtocol 80 Bulk-Only iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x88 EP 8 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0002 1x 2 bytes bInterval 9 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered Bus 001 Device 003: ID 064e:a219 Suyin Corp. 1.3M WebCam (notebook emachines E730, Acer sub-brand) Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x064e Suyin Corp. idProduct 0xa219 1.3M WebCam (notebook emachines E730, Acer sub-brand) bcdDevice 2.15 iManufacturer 2 Suyin iProduct 1 1.3M WebCam iSerial 3 HF1315-S32B-OV01-VA-R02.01.05 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 527 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 2 bFunctionClass 14 Video bFunctionSubClass 3 Video Interface Collection bFunctionProtocol 0 iFunction 5 1.3M WebCam Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 1 Video Control bInterfaceProtocol 0 iInterface 5 1.3M WebCam VideoControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdUVC 1.00 wTotalLength 77 dwClockFrequency 15.000000MHz bInCollection 1 baInterfaceNr( 0) 1 VideoControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 4 iTerminal 0 VideoControl Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 6 (EXTENSION_UNIT) bUnitID 4 guidExtensionCode {7033f028-1163-2e4a-ba2c- 6890eb334016} bNumControl 8 bNrPins 1 baSourceID( 0) 3 bControlSize 1 bmControls( 0) 0x0f iExtension 0 VideoControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Camera Sensor bAssocTerminal 0 iTerminal 0 wObjectiveFocalLengthMin 0 wObjectiveFocalLengthMax 0 wOcularFocalLength 0 bControlSize 3 bmControls 0x00000000 VideoControl Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 5 (PROCESSING_UNIT) Warning: Descriptor too short bUnitID 3 bSourceID 1 wMaxMultiplier 0 bControlSize 2 bmControls 0x0000153f Brightness Contrast Hue Saturation Sharpness Gamma Backlight Compensation Power Line Frequency White Balance Temperature, Auto iProcessing 0 bmVideoStandards 0xc6 NTSC - 525/60 PAL - 625/50 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 6 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 5 1.3M WebCam VideoStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 1 (INPUT_HEADER) bNumFormats 1 wTotalLength 307 bEndPointAddress 129 bmInfo 0 bTerminalLink 2 bStillCaptureMethod 2 bTriggerSupport 0 bTriggerUsage 0 bControlSize 1 bmaControls( 0) 27 VideoStreaming Interface Descriptor: bLength 27 bDescriptorType 36 bDescriptorSubtype 4 (FORMAT_UNCOMPRESSED) bFormatIndex 1 bNumFrameDescriptors 7 guidFormat {59555932-0000-1000-8000- 00aa00389b71} bBitsPerPixel 16 bDefaultFrameIndex 1 bAspectRatioX 0 bAspectRatioY 0 bmInterlaceFlags 0x00 Interlaced stream or variable: No Fields per frame: 2 fields Field 1 first: No Field pattern: Field 1 only bCopyProtect 0 VideoStreaming Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 1 bmCapabilities 0x00 Still image unsupported wWidth 640 wHeight 480 dwMinBitRate 73728000 dwMaxBitRate 147456000 dwMaxVideoFrameBufferSize 614400 dwDefaultFrameInterval 333333 bFrameIntervalType 2 dwFrameInterval( 0) 333333 dwFrameInterval( 1) 666666 VideoStreaming Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 2 bmCapabilities 0x00 Still image unsupported wWidth 352 wHeight 288 dwMinBitRate 24330240 dwMaxBitRate 48660480 dwMaxVideoFrameBufferSize 202752 dwDefaultFrameInterval 333333 bFrameIntervalType 2 dwFrameInterval( 0) 333333 dwFrameInterval( 1) 666666 VideoStreaming Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 3 bmCapabilities 0x00 Still image unsupported wWidth 320 wHeight 240 dwMinBitRate 18432000 dwMaxBitRate 36864000 dwMaxVideoFrameBufferSize 153600 dwDefaultFrameInterval 333333 bFrameIntervalType 2 dwFrameInterval( 0) 333333 dwFrameInterval( 1) 666666 VideoStreaming Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 4 bmCapabilities 0x00 Still image unsupported wWidth 176 wHeight 144 dwMinBitRate 6082560 dwMaxBitRate 12165120 dwMaxVideoFrameBufferSize 50688 dwDefaultFrameInterval 333333 bFrameIntervalType 2 dwFrameInterval( 0) 333333 dwFrameInterval( 1) 666666 VideoStreaming Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 5 bmCapabilities 0x00 Still image unsupported wWidth 160 wHeight 120 dwMinBitRate 4608000 dwMaxBitRate 9216000 dwMaxVideoFrameBufferSize 38400 dwDefaultFrameInterval 333333 bFrameIntervalType 2 dwFrameInterval( 0) 333333 dwFrameInterval( 1) 666666 VideoStreaming Interface Descriptor: bLength 30 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 6 bmCapabilities 0x00 Still image unsupported wWidth 1280 wHeight 800 dwMinBitRate 122880000 dwMaxBitRate 122880000 dwMaxVideoFrameBufferSize 2048000 dwDefaultFrameInterval 1333333 bFrameIntervalType 1 dwFrameInterval( 0) 1333333 VideoStreaming Interface Descriptor: bLength 30 bDescriptorType 36 bDescriptorSubtype 5 (FRAME_UNCOMPRESSED) bFrameIndex 7 bmCapabilities 0x00 Still image unsupported wWidth 1280 wHeight 1024 dwMinBitRate 157286400 dwMaxBitRate 157286400 dwMaxVideoFrameBufferSize 2621440 dwDefaultFrameInterval 1333333 bFrameIntervalType 1 dwFrameInterval( 0) 1333333 VideoStreaming Interface Descriptor: bLength 30 bDescriptorType 36 bDescriptorSubtype 3 (STILL_IMAGE_FRAME) bEndpointAddress 0 bNumImageSizePatterns 6 wWidth( 0) 1280 wHeight( 0) 1024 wWidth( 1) 640 wHeight( 1) 480 wWidth( 2) 320 wHeight( 2) 240 wWidth( 3) 320 wHeight( 3) 240 wWidth( 4) 176 wHeight( 4) 144 wWidth( 5) 160 wHeight( 5) 120 bNumCompressionPatterns 6 VideoStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 13 (COLORFORMAT) bColorPrimaries 1 (BT.709,sRGB) bTransferCharacteristics 1 (BT.709) bMatrixCoefficients 4 (SMPTE 170M (BT.601)) Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0080 1x 128 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0320 1x 800 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0b20 2x 800 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 5 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x1320 3x 800 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 6 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x1400 3x 1024 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0002 (Bus Powered) Remote Wakeup Enabled Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x8087 Intel Corp. idProduct 0x0020 Integrated Rate Matching Hub bcdDevice 0.00 iManufacturer 0 iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 6 wHubCharacteristic 0x0089 Per-port power switching Per-port overcurrent protection TT think time 8 FS bits Port indicators bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0xff Hub Port Status: Port 1: 0000.0503 highspeed power enable connect Port 2: 0000.0503 highspeed power enable connect Port 3: 0000.0100 power Port 4: 0000.0100 power Port 5: 0000.0100 power Port 6: 0000.0100 power Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 Full speed (or root) hub bMaxPacketSize0 64 idVendor 0x1d6b Linux Foundation idProduct 0x0002 2.0 root hub bcdDevice 5.00 iManufacturer 3 Linux 5.0.2 ehci_hcd iProduct 2 EHCI Host Controller iSerial 1 0000:00:1a.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 Full speed (or root) hub iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 3 wHubCharacteristic 0x000a No power switching (usb 1.0) Per-port overcurrent protection bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0xff Hub Port Status: Port 1: 0000.0503 highspeed power enable connect Port 2: 0000.0100 power Port 3: 0000.0100 power Device Status: 0x0001 Self Powered Does anybody know how to fix this problem ? Regards, Guido

3 20

[alsa-devel] Dell XPS 9360 continuous hissing sound
by Paulo Fidalgo 22 Mar '19

22 Mar '19

Hi! After a bug report on Fedora (https://bugzilla.redhat.com/show_bug.cgi?id=1677353) ---- When I start t he computer I hears a continuous background noise. The problem is also described here: https://wiki.archlinux.org/index.php/Dell_XPS_13_(9360)#Continuous_hissing_… I need to issue the command `amixer -c 0 cset 'numid=10' 1` to solve the problem, but every time I reboot the computer I need to run it again. This has been happening for quite some time, but I've been post-poning the bug report. Current kernel: 4.20.7-200.fc29.x86_64 Alsa information: http://www.alsa-project.org/db/?f=df4834c53335709abb521c767e7b1da2f4371744 ---- They advise me to contact you, to see if we can get a solution on this. Note: The computer also does not recognise (switch to) when I plug an headset. If you need more information, I'm eager to help diagnose and help to solve this, but I have no Kernel programming skills, other than that count on me. Kind regards, Paulo Fidalgo

2 3

[alsa-devel] [PATCH] ASoC: codecs: Fix to avoid potential NULL pointer dereferences
by Aditya Pakki 22 Mar '19

22 Mar '19

In sirf_audio_codec_driver_probe, of_match_node may fail and return a NULL pointer. The patch avoids a potential NULL pointer dereference. Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> --- sound/soc/codecs/sirf-audio-codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index e424499a8450..fd2bbd6a3c58 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -464,6 +464,8 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev) const struct of_device_id *match; match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); + if (!match) + return -ENXIO; sirf_audio_codec = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_codec), GFP_KERNEL); -- 2.17.1

2 2

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Simplify alt firmware loading code
by Takashi Iwai 22 Mar '19

22 Mar '19

ca0132 codec driver loads the firmware selectively depending on the model in addition to the fallback of the default firmware. The code works good, but a minor problem is that the current code seems confusing for Clang where it spews a warning about uninitialized variable. This patch simplifies the code flow for such a false-positive warning. After this refactoring, the ca0132_spec.alt_firmware_present field is no longer used, hence it's eliminated as well. Reported-and-tested-by: Arnd Bergmann <arnd(a)arndb.de> Reviewed-by: Nathan Chancellor <natechancellor(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 29882bda7632..e1ebc6d5f382 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1005,7 +1005,6 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; - bool alt_firmware_present; bool startup_check_entered; bool dsp_reload; @@ -7518,7 +7517,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) bool dsp_loaded = false; struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; /* * Alternate firmwares for different variants. The Recon3Di apparently * can use the default firmware, but I'll leave the option in case @@ -7529,33 +7528,26 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) case QUIRK_R3D: case QUIRK_AE5: if (request_firmware(&fw_entry, DESKTOP_EFX_FILE, - codec->card->dev) != 0) { + codec->card->dev) != 0) codec_dbg(codec, "Desktop firmware not found."); - spec->alt_firmware_present = false; - } else { + else codec_dbg(codec, "Desktop firmware selected."); - spec->alt_firmware_present = true; - } break; case QUIRK_R3DI: if (request_firmware(&fw_entry, R3DI_EFX_FILE, - codec->card->dev) != 0) { + codec->card->dev) != 0) codec_dbg(codec, "Recon3Di alt firmware not detected."); - spec->alt_firmware_present = false; - } else { + else codec_dbg(codec, "Recon3Di firmware selected."); - spec->alt_firmware_present = true; - } break; default: - spec->alt_firmware_present = false; break; } /* * Use default ctefx.bin if no alt firmware is detected, or if none * exists for your particular codec. */ - if (!spec->alt_firmware_present) { + if (!fw_entry) { codec_dbg(codec, "Default firmware selected."); if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: pcm: Fix possible OOB access in PCM oss plugins
by Takashi Iwai 22 Mar '19

22 Mar '19

The PCM OSS emulation converts and transfers the data on the fly via "plugins". The data is converted over the dynamically allocated buffer for each plugin, and recently syzkaller caught OOB in this flow. Although the bisection by syzbot pointed out to the commit 65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer allocations"), this is merely a commit to replace vmalloc() with kvmalloc(), hence it can't be the cause. The further debug action revealed that this happens in the case where a slave PCM doesn't support only the stereo channels while the OSS stream is set up for a mono channel. Below is a brief explanation: At each OSS parameter change, the driver sets up the PCM hw_params again in snd_pcm_oss_change_params_lock(). This is also the place where plugins are created and local buffers are allocated. The problem is that the plugins are created before the final hw_params is determined. Namely, two snd_pcm_hw_param_near() calls for setting the period size and periods may influence on the final result of channels, rates, etc, too, while the current code has already created plugins beforehand with the premature values. So, the plugin believes that channels=1, while the actual I/O is with channels=2, which makes the driver reading/writing over the allocated buffer size. The fix is simply to move the plugin allocation code after the final hw_params call. Reported-by: syzbot+d4503ae45b65c5bc1194(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 43 ++++++++++++++++++++++--------------------- 1 file changed, 22 insertions(+), 21 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d5b0d7ba83c4..f6ae68017608 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -940,6 +940,28 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) oss_frame_size = snd_pcm_format_physical_width(params_format(params)) * params_channels(params) / 8; + err = snd_pcm_oss_period_size(substream, params, sparams); + if (err < 0) + goto failure; + + n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size); + err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL); + if (err < 0) + goto failure; + + err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS, + runtime->oss.periods, NULL); + if (err < 0) + goto failure; + + snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); + + err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams); + if (err < 0) { + pcm_dbg(substream->pcm, "HW_PARAMS failed: %i\n", err); + goto failure; + } + #ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_oss_plugin_clear(substream); if (!direct) { @@ -974,27 +996,6 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) } #endif - err = snd_pcm_oss_period_size(substream, params, sparams); - if (err < 0) - goto failure; - - n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size); - err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL); - if (err < 0) - goto failure; - - err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS, - runtime->oss.periods, NULL); - if (err < 0) - goto failure; - - snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); - - if ((err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams)) < 0) { - pcm_dbg(substream->pcm, "HW_PARAMS failed: %i\n", err); - goto failure; - } - if (runtime->oss.trigger) { sw_params->start_threshold = 1; } else { -- 2.16.4

1 0

[alsa-devel] [PATCH] ALSA: hda/ca0132: work around clang -Wuninitialized warning
by Arnd Bergmann 22 Mar '19

22 Mar '19

When CONFIG_PCI is disabled, clang gets confused about the control flow of the switch() statement always ending up in the default case, and warns: sound/pci/hda/patch_ca0132.c:7558:6: error: variable 'fw_entry' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] if (!spec->alt_firmware_present) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/patch_ca0132.c:7565:42: note: uninitialized use occurs here dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); ^~~~~~~~ sound/pci/hda/patch_ca0132.c:7558:2: note: remove the 'if' if its condition is always true if (!spec->alt_firmware_present) { ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/patch_ca0132.c:7521:33: note: initialize the variable 'fw_entry' to silence this warning const struct firmware *fw_entry; ^ = NULL Adding an explicit check for CONFIG_PCI avoids the issue. Unfortunately this is not very intuitive here. Link: https://bugs.llvm.org/show_bug.cgi?id=41197#c1 Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- Any suggestions for other workarounds appreciated. If you can think of a better fix, please treat this as a reported-by: --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 29882bda7632..415b16b7db70 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7555,7 +7555,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) * Use default ctefx.bin if no alt firmware is detected, or if none * exists for your particular codec. */ - if (!spec->alt_firmware_present) { + if (!IS_ENABLED(CONFIG_PCI) || !spec->alt_firmware_present) { codec_dbg(codec, "Default firmware selected."); if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) -- 2.20.0

3 4

[alsa-devel] [PATCH 1/3] ALSA: hda/realtek: Enable ASUS X441MB and X705FD headset MIC with ALC256
by Jian-Hong Pan 22 Mar '19

22 Mar '19

The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 89a96442eb07..2523e495924e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5688,6 +5688,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6696,6 +6697,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7333,6 +7343,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x1a, 0x90a70130}, + {0x1b, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x12, 0xb7a60130}, {0x13, 0xb8a61140}, -- 2.20.1

2 5

[alsa-devel] [PATCH] ASoC: AMD: Configure wclk and bclk of master codec
by Agrawal, Akshu 22 Mar '19

22 Mar '19

With CCF support in da7219, we can now set the correct rate of wclk and bclk. Setting bclk at lower rate at 1.53Mhz from its earlier default rate of 3Mhz, also fixes noise issue observed on some dmics. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++------- 1 file changed, 30 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f37a588ba345..1bb5b78b6e10 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -46,7 +46,8 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -static struct clk *da7219_dai_clk; +static struct clk *da7219_dai_wclk; +static struct clk *da7219_dai_bclk; extern int bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -72,7 +73,8 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -94,12 +96,15 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream) +static int da7219_clk_enable(struct snd_pcm_substream *substream, + int wclk_rate, int bclk_rate) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - ret = clk_prepare_enable(da7219_dai_clk); + clk_set_rate(da7219_dai_wclk, wclk_rate); + clk_set_rate(da7219_dai_bclk, bclk_rate); + ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); return ret; @@ -110,7 +115,7 @@ static int da7219_clk_enable(struct snd_pcm_substream *substream) static void da7219_clk_disable(void) { - clk_disable_unprepare(da7219_dai_clk); + clk_disable_unprepare(da7219_dai_bclk); } static const unsigned int channels[] = { @@ -151,7 +156,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -173,7 +178,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return da7219_clk_enable(substream); + return 0; } static void cz_da7219_shutdown(struct snd_pcm_substream *substream) @@ -199,7 +204,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static void cz_max_shutdown(struct snd_pcm_substream *substream) @@ -225,7 +230,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -247,7 +252,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + return 0; } static void cz_dmic_shutdown(struct snd_pcm_substream *substream) @@ -255,29 +260,44 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int wclk, bclk; + + wclk = params_rate(params); + bclk = wclk * params_channels(params) * + snd_pcm_format_width(params_format(params)); + return da7219_clk_enable(substream, wclk, bclk); +} static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

1 0

[alsa-devel] [asoc:for-5.2 14/67] nau8810.c:undefined reference to `__udivdi3'
by kbuild test robot 22 Mar '19

22 Mar '19

Hi John, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 008fe4e5382293e2da96a4dc75b95782a5f95be8 commit: b517229ca2f7836125be58997808f2803f9ebc86 [14/67] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit config: m68k-allyesconfig (attached as .config) compiler: m68k-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 b517229ca2f7836125be58997808f2803f9ebc86 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=m68k All errors (new ones prefixed by >>): sound/soc/codecs/nau8810.o: In function `nau8810_set_pll': >> nau8810.c:(.text+0x53c): undefined reference to `__udivdi3' --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: fix hdmi codec driver contest in S3
by libin.yang＠intel.com 22 Mar '19

22 Mar '19

From: Libin Yang <libin.yang(a)intel.com> In S3, there is a contest between dapm event and getting display power. When resume, the sequence is: hdac_hdmi_runtime_resume() => snd_hdac_display_power() hdac_hdmi_xxx_widget_event() The hdac_hdmi_xxx_widget_event() are based on the power being on. Otherwise, the operation on the codec register will fail. However, in snd_hdac_display_power(), it may sleep because of the mutex_lock() in display driver. In this case, hdac_hdmi_xxx_widget_event() will be called before the power is on. Let's not operate the registers and wait for the power in hdac_hdmi_xxx_widget_event() Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..dc182b0 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -144,6 +144,7 @@ struct hdac_hdmi_priv { int num_pin; int num_cvt; int num_ports; + int display_power; /* 0: power off; 1 power on */ struct mutex pin_mutex; struct hdac_chmap chmap; struct hdac_hdmi_drv_data *drv_data; @@ -742,12 +743,26 @@ static void hdac_hdmi_set_amp(struct hdac_device *hdev, AC_VERB_SET_AMP_GAIN_MUTE, val); } +static int wait_for_display_power(struct hdac_hdmi_priv *hdmi) +{ + int i = 0; + + /* sleep for totally 80ms ~ 200ms should be enough */ + while (i < 40) { + if (!hdmi->display_power) + usleep_range(2000, 5000); + else + return 0; + } + return -EAGAIN; +} static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct hdac_hdmi_port *port = w->priv; struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm; dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", @@ -757,6 +772,11 @@ static int hdac_hdmi_pin_output_widget_event(struct snd_soc_dapm_widget *w, if (!pcm) return -EIO; + if (wait_for_display_power(hdmi)) { + dev_err(&hdev->dev, "%s: hdmi power is not ready\n", __func__); + return -EAGAIN; + } + /* set the device if pin is mst_capable */ if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; @@ -803,6 +823,11 @@ static int hdac_hdmi_cvt_output_widget_event(struct snd_soc_dapm_widget *w, if (!pcm) return -EIO; + if (wait_for_display_power(hdmi)) { + dev_err(&hdev->dev, "%s: hdmi power is not ready\n", __func__); + return -EAGAIN; + } + switch (event) { case SND_SOC_DAPM_PRE_PMU: hdac_hdmi_set_power_state(hdev, cvt->nid, AC_PWRST_D0); @@ -840,6 +865,7 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, { struct hdac_hdmi_port *port = w->priv; struct hdac_device *hdev = dev_to_hdac_dev(w->dapm->dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); int mux_idx; dev_dbg(&hdev->dev, "%s: widget: %s event: %x\n", @@ -850,6 +876,11 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, mux_idx = dapm_kcontrol_get_value(kc); + if (wait_for_display_power(hdmi)) { + dev_err(&hdev->dev, "%s: hdmi power is not ready\n", __func__); + return -EAGAIN; + } + /* set the device if pin is mst_capable */ if (hdac_hdmi_port_select_set(hdev, port) < 0) return -EIO; @@ -2068,6 +2099,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); struct hdac_bus *bus = hdev->bus; + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_ext_link *hlink = NULL; dev_dbg(dev, "Enter: %s\n", __func__); @@ -2095,6 +2127,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(bus, hlink); snd_hdac_display_power(bus, hdev->addr, false); + hdmi->display_power = 0; return 0; } @@ -2102,6 +2135,7 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) static int hdac_hdmi_runtime_resume(struct device *dev) { struct hdac_device *hdev = dev_to_hdac_dev(dev); + struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_bus *bus = hdev->bus; struct hdac_ext_link *hlink = NULL; @@ -2128,6 +2162,7 @@ static int hdac_hdmi_runtime_resume(struct device *dev) snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + hdmi->display_power = 1; return 0; } #else -- 2.7.4

1 0

[alsa-devel] [PATCH v2 0/7] ASoC: tlv320aic26: Add device tree support and bug fixes
by Cormier, Jonathan 21 Mar '19

21 Mar '19

Hi, The following patches were created to get the tlv320aic26 working on our AM335x SoM. I'm resubmitting this patch series. Previous version here: https://lore.kernel.org/patchwork/cover/601439/ This patch series relies on the following patch for proper operation. commit 93d0ad8f374c ("ASoC: tlv320aic26: Convert to params_width()") Changes since RFC: - Removed the unnecessary "#if defined(CONFIG_OF)" - Dropped the DSP_B patch Changes since V1: - Reordered Kconfig patch to first so each patch can be built - Rebased changes to Linux 5.1-rc1 Cormier, Jonathan (7): ASoC: tlv320aic26: Allow tlv320aic26 to be selected in menuconfig dt-bindings: ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Fix module autoload ASoC: tlv320aic26: Fix regmap by setting reg_defaults and reg_stride ASoC: tlv320aic26: hw_params was unintentionally clearing AIC26 master mode ASoC: tlv320aic26: Change Capture Mute to Capture Switch to match alsa documentation. .../devicetree/bindings/sound/tlv320aic26.txt | 65 +++++++++++++++++++ sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/tlv320aic26.c | 55 ++++++++++++++-- 3 files changed, 116 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic26.txt -- 2.21.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix NULL pointer dereference in snd_soc_dapm_new_dai
by Pankaj Bharadiya 21 Mar '19

21 Mar '19

In case of single config, w_param_text is NULL. In snd_soc_dapm_new_control_unlocked() call failure case, it will end up calling snd_soc_dapm_free_kcontrol() unconditionally and result in NULL pointer dereference. Signed-off-by: Pankaj Bharadiya <pankaj.laxminarayan.bharadiya(a)intel.com> --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ec06ef..ba6cb37 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4094,8 +4094,9 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); - snd_soc_dapm_free_kcontrol(card, &private_value, - rtd->dai_link->num_params, w_param_text); + if (w_param_text) + snd_soc_dapm_free_kcontrol(card, &private_value, + rtd->dai_link->num_params, w_param_text); param_fail: devm_kfree(card->dev, link_name); return ERR_PTR(ret); -- 2.7.4

4 4

[alsa-devel] [PATCH] ASoC: cs4270: Set auto-increment bit for register writes
by Daniel Mack 21 Mar '19

21 Mar '19

The CS4270 does not by default increment the register address on consecutive writes. During normal operation it doesn't matter as all register accesses are done individually. At resume time after suspend, however, the regcache code gathers the biggest possible block of registers to sync and sends them one on one go. To fix this, set the INCR bit in all cases. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/cs4270.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3c266eeb89bf..007ce9f48e44 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -642,6 +642,7 @@ static const struct regmap_config cs4270_regmap = { .reg_defaults = cs4270_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4270_reg_defaults), .cache_type = REGCACHE_RBTREE, + .write_flag_mask = CS4270_I2C_INCR, .readable_reg = cs4270_reg_is_readable, .volatile_reg = cs4270_reg_is_volatile, -- 2.20.1

2 1

[alsa-devel] [PATCH V3] ASoC: fsl_asrc: add constraint for the asrc of older version
by S.j. Wang 21 Mar '19

21 Mar '19

There is a constraint for the channel number setting on the asrc of older version (e.g. imx35), the channel number should be even, odd number isn't valid. So add this constraint when the asrc of older version is used. Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v3 - fix comments with Nicolin's review sound/soc/fsl/fsl_asrc.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 528e8b108422..0b937924d2e4 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -445,6 +445,19 @@ struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir) } EXPORT_SYMBOL_GPL(fsl_asrc_get_dma_channel); +static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); + + /* Odd channel number is not valid for older ASRC (channel_bits==3) */ + if (asrc_priv->channel_bits == 3) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, 2); + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -539,6 +552,7 @@ static int fsl_asrc_dai_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops fsl_asrc_dai_ops = { + .startup = fsl_asrc_dai_startup, .hw_params = fsl_asrc_dai_hw_params, .hw_free = fsl_asrc_dai_hw_free, .trigger = fsl_asrc_dai_trigger, -- 1.9.1

2 1

[alsa-devel] [PATCH V4] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 21 Mar '19

21 Mar '19

There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the beginning with the normal enable flow. This is hardware issue, but there is no errata, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting. Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 47 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 37 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..3623aa9a6f2e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -54,6 +54,8 @@ struct fsl_esai { u32 fifo_depth; u32 slot_width; u32 slots; + u32 tx_mask; + u32 rx_mask; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -361,21 +363,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); - esai_priv->slot_width = slot_width; esai_priv->slots = slots; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -596,6 +590,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 mask; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -608,15 +603,38 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, for (i = 0; tx && i < channels; i++) regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + /* + * When set the TE/RE in the end of enablement flow, there + * will be channel swap issue for multi data line case. + * In order to workaround this issue, we switch the bit + * enablement sequence to below sequence + * 1) clear the xSMB & xSMA: which is done in probe and + * stop state. + * 2) set TE/RE + * 3) set xSMB + * 4) set xSMA: xSMA is the last one in this flow, which + * will trigger esai to start. + */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); + mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, 0); /* Disable and reset FIFO */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), @@ -906,6 +924,15 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + esai_priv->tx_mask = 0xFFFFFFFF; + esai_priv->rx_mask = 0xFFFFFFFF; + + /* Clear the TSMA, TSMB, RSMA, RSMB */ + regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { -- 1.9.1

2 3

[alsa-devel] [PATCH 0/3] ASoC: da7219: Extend CCF support to allow WCLK/BCLK config
by Adam Thomson 21 Mar '19

21 Mar '19

This patch set extends the existing CCF support in the DA7219 codec driver by splitting the DAI clock gate into clocks to cover WCLK and BCLK, to enable the configuration of these clocks from audio machine drivers when this codec is the clock master of the DAI. Adam Thomson (3): ASoC: da7219: Update DAI clock binding info to cover WCLK/BCLK ASoC: da7219: Expose BCLK and WCLK control through CCF ASoC: AMD: Update DA7219 DAI clock name to align with codec updates Documentation/devicetree/bindings/sound/da7219.txt | 8 +- include/sound/da7219.h | 8 +- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/codecs/da7219.c | 523 ++++++++++++++++----- sound/soc/codecs/da7219.h | 6 +- 5 files changed, 412 insertions(+), 135 deletions(-) -- 1.9.1

2 8

[alsa-devel] Applied "ASoC: simple-card-utils: add asoc_simple_debug_info()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: add asoc_simple_debug_info() 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 0580dde59438686d60762b6da9229ebec693b94f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:54:42 +0900 Subject: [PATCH] ASoC: simple-card-utils: add asoc_simple_debug_info() Current simple-card-utils has dev_dbg(), but people want to add #define DEBUG at simple-card/audio-graph, not simple-card-utils. And, people want to get all information. This patch adds new asoc_simple_debug_info() to indicates information. 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 | 72 +++++++++++++++++++++++++++ sound/soc/generic/audio-graph-card.c | 2 + sound/soc/generic/simple-card-utils.c | 12 ----- sound/soc/generic/simple-card.c | 2 + 4 files changed, 76 insertions(+), 12 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 7afe45389972..0199ad0bbcd0 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -129,4 +129,76 @@ int asoc_simple_card_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix); + +#ifdef DEBUG +inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv, + char *name, + struct asoc_simple_dai *dai) +{ + struct device *dev = simple_priv_to_dev(priv); + + if (dai->name) + dev_dbg(dev, "%s dai name = %s\n", + name, dai->name); + if (dai->sysclk) + dev_dbg(dev, "%s sysclk = %d\n", + name, dai->sysclk); + + dev_dbg(dev, "%s direction = %s\n", + name, dai->clk_direction ? "OUT" : "IN"); + + if (dai->slots) + dev_dbg(dev, "%s slots = %d\n", name, dai->slots); + if (dai->slot_width) + dev_dbg(dev, "%s slot width = %d\n", name, dai->slot_width); + if (dai->tx_slot_mask) + dev_dbg(dev, "%s tx slot mask = %d\n", name, dai->tx_slot_mask); + if (dai->rx_slot_mask) + dev_dbg(dev, "%s rx slot mask = %d\n", name, dai->rx_slot_mask); + if (dai->clk) + dev_dbg(dev, "%s clk %luHz\n", name, clk_get_rate(dai->clk)); +} + +inline void asoc_simple_debug_info(struct asoc_simple_priv *priv) +{ + struct snd_soc_card *card = simple_priv_to_card(priv); + struct device *dev = simple_priv_to_dev(priv); + + int i; + + if (card->name) + dev_dbg(dev, "Card Name: %s\n", card->name); + + for (i = 0; i < card->num_links; i++) { + struct simple_dai_props *props = simple_priv_to_props(priv, i); + struct snd_soc_dai_link *link = simple_priv_to_link(priv, i); + + dev_dbg(dev, "DAI%d\n", i); + + asoc_simple_debug_dai(priv, "cpu", props->cpu_dai); + asoc_simple_debug_dai(priv, "codec", props->codec_dai); + + if (link->name) + dev_dbg(dev, "dai name = %s\n", link->name); + + dev_dbg(dev, "dai format = %04x\n", link->dai_fmt); + + if (props->adata.convert_rate) + dev_dbg(dev, "convert_rate = %d\n", + props->adata.convert_rate); + if (props->adata.convert_channels) + dev_dbg(dev, "convert_channels = %d\n", + props->adata.convert_channels); + if (props->codec_conf && props->codec_conf->name_prefix) + dev_dbg(dev, "name prefix = %s\n", + props->codec_conf->name_prefix); + if (props->mclk_fs) + dev_dbg(dev, "mclk-fs = %d\n", + props->mclk_fs); + } +} +#else +#define asoc_simple_debug_info(priv) +#endif /* DEBUG */ + #endif /* __SIMPLE_CARD_UTILS_H */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 7dd772a895e2..2a95355540d1 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -712,6 +712,8 @@ static int graph_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); + asoc_simple_debug_info(priv); + ret = devm_snd_soc_register_card(dev, card); if (ret < 0) goto err; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 5c1424f03620..f5ba5ad300cc 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -49,9 +49,6 @@ void asoc_simple_card_parse_convert(struct device *dev, /* channels transfer */ snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); of_property_read_u32(np, prop, &data->convert_channels); - - dev_dbg(dev, "convert_rate %d\n", data->convert_rate); - dev_dbg(dev, "convert_channels %d\n", data->convert_channels); } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert); @@ -93,8 +90,6 @@ int asoc_simple_card_parse_daifmt(struct device *dev, *retfmt = daifmt; - dev_dbg(dev, "format : %04x\n", daifmt); - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_daifmt); @@ -116,8 +111,6 @@ int asoc_simple_card_set_dailink_name(struct device *dev, dai_link->name = name; dai_link->stream_name = name; - - dev_dbg(dev, "name : %s\n", name); } return ret; @@ -146,8 +139,6 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, if (!card->name && card->dai_link) card->name = card->dai_link->name; - dev_dbg(card->dev, "Card Name: %s\n", card->name ? card->name : ""); - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); @@ -211,9 +202,6 @@ 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", dai_name, - simple_dai->sysclk, simple_dai->clk_direction); - return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b792638a11c8..77899624787e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -813,6 +813,8 @@ static int simple_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); + asoc_simple_debug_info(priv); + ret = devm_snd_soc_register_card(dev, card); if (ret < 0) goto err; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple_card_utils: share common priv for simple-card/audio-graph" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple_card_utils: share common priv for simple-card/audio-graph 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 e59289cda8dec0153fa396864c8ba8092ec3b80d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:54:59 +0900 Subject: [PATCH] ASoC: simple_card_utils: share common priv for simple-card/audio-graph Historically, simple-card/simple-scu-card/audio-graph/audio-graph-scu are similar but different generic sound card. simple-scu-card which was for DPCM was merged into simple-card, and audio-graph-scu which was for DPCM was merged into audio-graph. simple-card is for non OF graph sound card, and audio-graph is for OF graph sound card. And, small detail difference (= function parameter, naming, etc) between simple-card/audio-graph has been unified. So today, the difference between simple-card/audio-graph are just using OF graph style, or not. In other words, there should no difference other than OF graph sytle. simple-card/audio-graph are using own priv today , but we can merge it. This patch merge it at simple_card_utils. 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 | 23 +++++++ sound/soc/generic/audio-graph-card.c | 96 +++++++++++----------------- sound/soc/generic/simple-card.c | 57 +++++------------ 3 files changed, 76 insertions(+), 100 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0199ad0bbcd0..c0161ec17572 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -37,6 +37,29 @@ struct asoc_simple_jack { struct snd_soc_jack_gpio gpio; }; +struct asoc_simple_priv { + struct snd_soc_card snd_card; + struct simple_dai_props { + 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 platforms; + struct asoc_simple_card_data adata; + struct snd_soc_codec_conf *codec_conf; + unsigned int mclk_fs; + } *dai_props; + struct asoc_simple_jack hp_jack; + struct asoc_simple_jack mic_jack; + struct snd_soc_dai_link *dai_link; + struct asoc_simple_dai *dais; + struct snd_soc_codec_conf *codec_conf; + struct gpio_desc *pa_gpio; +}; +#define simple_priv_to_card(priv) (&(priv)->snd_card) +#define simple_priv_to_props(priv, i) ((priv)->dai_props + (i)) +#define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) +#define simple_priv_to_link(priv, i) (simple_priv_to_card(priv)->dai_link + (i)) + int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *node, struct device_node *codec, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 2a95355540d1..656ca49ba727 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -20,25 +20,6 @@ #include <linux/string.h> #include <sound/simple_card_utils.h> -struct graph_priv { - struct snd_soc_card snd_card; - 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 */ - struct snd_soc_dai_link_component platforms; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - unsigned int mclk_fs; - } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct snd_soc_codec_conf *codec_conf; - struct gpio_desc *pa_gpio; -}; - struct link_info { int dais; /* number of dai */ int link; /* number of link */ @@ -46,11 +27,6 @@ struct link_info { int cpu; /* turn for CPU / Codec */ }; -#define graph_priv_to_card(priv) (&(priv)->snd_card) -#define graph_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define graph_priv_to_dev(priv) (graph_priv_to_card(priv)->dev) -#define graph_priv_to_link(priv, i) (graph_priv_to_card(priv)->dai_link + (i)) - #define PREFIX "audio-graph-card," static int graph_outdrv_event(struct snd_soc_dapm_widget *w, @@ -58,7 +34,7 @@ static int graph_outdrv_event(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_dapm_context *dapm = w->dapm; - struct graph_priv *priv = snd_soc_card_get_drvdata(dapm->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(dapm->card); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -83,8 +59,8 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { static int graph_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); @@ -101,8 +77,8 @@ static int graph_startup(struct snd_pcm_substream *substream) static void graph_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_clk_disable(dai_props->cpu_dai); @@ -115,8 +91,8 @@ static int graph_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; int ret = 0; @@ -148,8 +124,8 @@ static const struct snd_soc_ops graph_ops = { static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret = 0; ret = asoc_simple_card_init_dai(rtd->codec_dai, @@ -168,8 +144,8 @@ static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct graph_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, rtd->num); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_convert_fixup(&dai_props->adata, params); @@ -198,7 +174,7 @@ static void graph_parse_convert(struct device *dev, static void graph_parse_mclk_fs(struct device_node *top, struct device_node *ep, - struct graph_dai_props *props) + struct simple_dai_props *props) { struct device_node *port = of_get_parent(ep); struct device_node *ports = of_get_parent(port); @@ -214,15 +190,15 @@ static void graph_parse_mclk_fs(struct device_node *top, of_node_put(node); } -static int graph_dai_link_of_dpcm(struct graph_priv *priv, +static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec) { - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct device_node *ep = li->cpu ? cpu_ep : codec_ep; struct device_node *port; @@ -344,14 +320,14 @@ static int graph_dai_link_of_dpcm(struct graph_priv *priv, return 0; } -static int graph_dai_link_of(struct graph_priv *priv, +static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li) { - struct device *dev = graph_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = graph_priv_to_link(priv, li->link); - struct graph_dai_props *dai_props = graph_priv_to_props(priv, li->link); + struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; @@ -420,19 +396,19 @@ static int graph_dai_link_of(struct graph_priv *priv, return 0; } -static int graph_for_each_link(struct graph_priv *priv, +static int graph_for_each_link(struct asoc_simple_priv *priv, struct link_info *li, - int (*func_noml)(struct graph_priv *priv, + int (*func_noml)(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li), - int (*func_dpcm)(struct graph_priv *priv, + int (*func_dpcm)(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec)) { struct of_phandle_iterator it; - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); struct device_node *node = dev->of_node; struct device_node *cpu_port; struct device_node *cpu_ep; @@ -488,9 +464,9 @@ static int graph_for_each_link(struct graph_priv *priv, return 0; } -static int graph_parse_of(struct graph_priv *priv) +static int graph_parse_of(struct asoc_simple_priv *priv) { - struct snd_soc_card *card = graph_priv_to_card(priv); + struct snd_soc_card *card = simple_priv_to_card(priv); struct link_info li; int ret; @@ -526,12 +502,12 @@ static int graph_parse_of(struct graph_priv *priv) return asoc_simple_card_parse_card_name(card, NULL); } -static int graph_count_noml(struct graph_priv *priv, +static int graph_count_noml(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li) { - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); li->link += 1; /* 1xCPU-Codec */ li->dais += 2; /* 1xCPU + 1xCodec */ @@ -541,13 +517,13 @@ static int graph_count_noml(struct graph_priv *priv, return 0; } -static int graph_count_dpcm(struct graph_priv *priv, +static int graph_count_dpcm(struct asoc_simple_priv *priv, struct device_node *cpu_ep, struct device_node *codec_ep, struct link_info *li, int dup_codec) { - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); li->link++; /* 1xCPU-dummy */ li->dais++; /* 1xCPU */ @@ -563,10 +539,10 @@ static int graph_count_dpcm(struct graph_priv *priv, return 0; } -static void graph_get_dais_count(struct graph_priv *priv, +static void graph_get_dais_count(struct asoc_simple_priv *priv, struct link_info *li) { - struct device *dev = graph_priv_to_dev(priv); + struct device *dev = simple_priv_to_dev(priv); /* * link_num : number of links. @@ -623,7 +599,7 @@ static void graph_get_dais_count(struct graph_priv *priv, static int graph_card_probe(struct snd_soc_card *card) { - struct graph_priv *priv = snd_soc_card_get_drvdata(card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); @@ -639,9 +615,9 @@ static int graph_card_probe(struct snd_soc_card *card) static int graph_probe(struct platform_device *pdev) { - struct graph_priv *priv; + struct asoc_simple_priv *priv; struct snd_soc_dai_link *dai_link; - struct graph_dai_props *dai_props; + struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; struct device *dev = &pdev->dev; struct snd_soc_card *card; @@ -654,7 +630,7 @@ static int graph_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - card = graph_priv_to_card(priv); + card = simple_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; card->dapm_widgets = graph_dapm_widgets; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 77899624787e..628c19a3aaff 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -15,24 +15,6 @@ #include <sound/soc-dai.h> #include <sound/soc.h> -struct simple_priv { - struct snd_soc_card snd_card; - struct simple_dai_props { - 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 platforms; - struct asoc_simple_card_data adata; - struct snd_soc_codec_conf *codec_conf; - unsigned int mclk_fs; - } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; - struct snd_soc_dai_link *dai_link; - struct asoc_simple_dai *dais; - struct snd_soc_codec_conf *codec_conf; -}; - struct link_info { int dais; /* number of dai */ int link; /* number of link */ @@ -40,11 +22,6 @@ struct link_info { int cpu; /* turn for CPU / Codec */ }; -#define simple_priv_to_card(priv) (&(priv)->snd_card) -#define simple_priv_to_props(priv, i) ((priv)->dai_props + (i)) -#define simple_priv_to_dev(priv) (simple_priv_to_card(priv)->dev) -#define simple_priv_to_link(priv, i) (simple_priv_to_card(priv)->dai_link + (i)) - #define DAI "sound-dai" #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," @@ -52,7 +29,7 @@ struct link_info { static int simple_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; @@ -71,7 +48,7 @@ static int simple_startup(struct snd_pcm_substream *substream) static void simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); @@ -101,7 +78,7 @@ static int simple_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; @@ -144,7 +121,7 @@ static const struct snd_soc_ops simple_ops = { static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; @@ -164,7 +141,7 @@ static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - struct simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); asoc_simple_card_convert_fixup(&dai_props->adata, params); @@ -207,7 +184,7 @@ static void simple_parse_mclk_fs(struct device_node *top, of_node_put(node); } -static int simple_dai_link_of_dpcm(struct simple_priv *priv, +static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, @@ -336,7 +313,7 @@ static int simple_dai_link_of_dpcm(struct simple_priv *priv, return 0; } -static int simple_dai_link_of(struct simple_priv *priv, +static int simple_dai_link_of(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, @@ -438,13 +415,13 @@ static int simple_dai_link_of(struct simple_priv *priv, return ret; } -static int simple_for_each_link(struct simple_priv *priv, +static int simple_for_each_link(struct asoc_simple_priv *priv, struct link_info *li, - int (*func_noml)(struct simple_priv *priv, + int (*func_noml)(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top), - int (*func_dpcm)(struct simple_priv *priv, + int (*func_dpcm)(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top)) @@ -513,7 +490,7 @@ static int simple_for_each_link(struct simple_priv *priv, } static int simple_parse_aux_devs(struct device_node *node, - struct simple_priv *priv) + struct asoc_simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *aux_node; @@ -543,7 +520,7 @@ static int simple_parse_aux_devs(struct device_node *node, return 0; } -static int simple_parse_of(struct simple_priv *priv) +static int simple_parse_of(struct asoc_simple_priv *priv) { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; @@ -593,7 +570,7 @@ static int simple_parse_of(struct simple_priv *priv) return ret; } -static int simple_count_noml(struct simple_priv *priv, +static int simple_count_noml(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top) @@ -605,7 +582,7 @@ static int simple_count_noml(struct simple_priv *priv, return 0; } -static int simple_count_dpcm(struct simple_priv *priv, +static int simple_count_dpcm(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, struct link_info *li, bool is_top) @@ -618,7 +595,7 @@ static int simple_count_dpcm(struct simple_priv *priv, return 0; } -static void simple_get_dais_count(struct simple_priv *priv, +static void simple_get_dais_count(struct asoc_simple_priv *priv, struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); @@ -687,7 +664,7 @@ static void simple_get_dais_count(struct simple_priv *priv, static int simple_soc_probe(struct snd_soc_card *card) { - struct simple_priv *priv = snd_soc_card_get_drvdata(card); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); @@ -703,7 +680,7 @@ static int simple_soc_probe(struct snd_soc_card *card) static int simple_probe(struct platform_device *pdev) { - struct simple_priv *priv; + struct asoc_simple_priv *priv; struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: share asoc_simple_startup()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: share asoc_simple_startup() 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 f38df5bf0c9cb905fa9d5abc86c3a00128cdbba5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:55:14 +0900 Subject: [PATCH] ASoC: simple-card-utils: share asoc_simple_startup() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_startup() between in these 2 drivers. 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 | 2 +- sound/soc/generic/audio-graph-card.c | 20 +------------------- sound/soc/generic/simple-card-utils.c | 22 ++++++++++++++++++++-- sound/soc/generic/simple-card.c | 21 +-------------------- 4 files changed, 23 insertions(+), 42 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index c0161ec17572..0653fb9385d1 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -84,8 +84,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct asoc_simple_dai *simple_dai, 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); +int asoc_simple_startup(struct snd_pcm_substream *substream); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 656ca49ba727..29ae71b427ec 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -56,24 +56,6 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static int graph_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - static void graph_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -117,7 +99,7 @@ static int graph_hw_params(struct snd_pcm_substream *substream, } static const struct snd_soc_ops graph_ops = { - .startup = graph_startup, + .startup = asoc_simple_startup, .shutdown = graph_shutdown, .hw_params = graph_hw_params, }; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index f5ba5ad300cc..83ea1afa3c72 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -143,14 +143,13 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); -int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) +static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) { if (dai) return clk_prepare_enable(dai->clk); return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_clk_enable); void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { @@ -206,6 +205,25 @@ int asoc_simple_card_parse_clk(struct device *dev, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); +int asoc_simple_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + if (ret) + return ret; + + ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + if (ret) + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + return ret; +} +EXPORT_SYMBOL_GPL(asoc_simple_startup); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 628c19a3aaff..d0ea5736889e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -26,25 +26,6 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static int simple_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); - if (ret) - return ret; - - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); - if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - return ret; -} - static void simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -114,7 +95,7 @@ static int simple_hw_params(struct snd_pcm_substream *substream, } static const struct snd_soc_ops simple_ops = { - .startup = simple_startup, + .startup = asoc_simple_startup, .shutdown = simple_shutdown, .hw_params = simple_hw_params, }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: share asoc_simple_shutdown()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: share asoc_simple_shutdown() 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 686911b46fb5a08df142fe22b6c06dc6fbd3ba65 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:55:27 +0900 Subject: [PATCH] ASoC: simple-card-utils: share asoc_simple_shutdown() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_shutdown() between in these 2 drivers. 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 | 2 +- sound/soc/generic/audio-graph-card.c | 13 +------------ sound/soc/generic/simple-card-utils.c | 16 ++++++++++++++-- sound/soc/generic/simple-card.c | 14 +------------- 4 files changed, 17 insertions(+), 28 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0653fb9385d1..edda646c3ebe 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -84,8 +84,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct asoc_simple_dai *simple_dai, const char *dai_name, struct snd_soc_dai_link_component *dlc); -void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); int asoc_simple_startup(struct snd_pcm_substream *substream); +void asoc_simple_shutdown(struct snd_pcm_substream *substream); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 29ae71b427ec..f3577b5bf14c 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -56,17 +56,6 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static void graph_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - static int graph_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -100,7 +89,7 @@ static int graph_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops graph_ops = { .startup = asoc_simple_startup, - .shutdown = graph_shutdown, + .shutdown = asoc_simple_shutdown, .hw_params = graph_hw_params, }; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 83ea1afa3c72..7ede16347ea8 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -151,12 +151,11 @@ static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) return 0; } -void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) +static void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) { if (dai) clk_disable_unprepare(dai->clk); } -EXPORT_SYMBOL_GPL(asoc_simple_card_clk_disable); int asoc_simple_card_parse_clk(struct device *dev, struct device_node *node, @@ -224,6 +223,19 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(asoc_simple_startup); +void asoc_simple_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + simple_priv_to_props(priv, rtd->num); + + asoc_simple_card_clk_disable(dai_props->cpu_dai); + + asoc_simple_card_clk_disable(dai_props->codec_dai); +} +EXPORT_SYMBOL_GPL(asoc_simple_shutdown); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d0ea5736889e..bb57c482eac8 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -26,18 +26,6 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static void simple_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_clk_disable(dai_props->cpu_dai); - - asoc_simple_card_clk_disable(dai_props->codec_dai); -} - static int simple_set_clk_rate(struct asoc_simple_dai *simple_dai, unsigned long rate) { @@ -96,7 +84,7 @@ static int simple_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops simple_ops = { .startup = asoc_simple_startup, - .shutdown = simple_shutdown, + .shutdown = asoc_simple_shutdown, .hw_params = simple_hw_params, }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: share asoc_simple_hw_param()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: share asoc_simple_hw_param() 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 f48dcbb6d47d870cf3a03f453c923dd262158c66 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:55:39 +0900 Subject: [PATCH] ASoC: simple-card-utils: share asoc_simple_hw_param() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_hw_param() between in these 2 drivers. One note is that only simple-card supports simple_set_clk_rate() at hw_param from commit e9be4ffd4f40fcb ("ASoC: simple-card: set cpu dai clk in hw_params"). By this patch, audio-graph has same feature. 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 | 2 + sound/soc/generic/audio-graph-card.c | 33 +-------------- sound/soc/generic/simple-card-utils.c | 57 ++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 58 +-------------------------- 4 files changed, 61 insertions(+), 89 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index edda646c3ebe..89ae0841b711 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -86,6 +86,8 @@ int asoc_simple_card_parse_clk(struct device *dev, struct snd_soc_dai_link_component *dlc); int asoc_simple_startup(struct snd_pcm_substream *substream); void asoc_simple_shutdown(struct snd_pcm_substream *substream); +int asoc_simple_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index f3577b5bf14c..51160adaf5bb 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -56,41 +56,10 @@ static const struct snd_soc_dapm_widget graph_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), }; -static int graph_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - unsigned int mclk, mclk_fs = 0; - int ret = 0; - - if (dai_props->mclk_fs) - mclk_fs = dai_props->mclk_fs; - - if (mclk_fs) { - mclk = params_rate(params) * mclk_fs; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) - goto err; - - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret && ret != -ENOTSUPP) - goto err; - } - return 0; -err: - return ret; -} - static const struct snd_soc_ops graph_ops = { .startup = asoc_simple_startup, .shutdown = asoc_simple_shutdown, - .hw_params = graph_hw_params, + .hw_params = asoc_simple_hw_params, }; static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 7ede16347ea8..de23cf8bffbd 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -236,6 +236,63 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(asoc_simple_shutdown); +static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, + unsigned long rate) +{ + if (!simple_dai) + return 0; + + if (!simple_dai->clk) + return 0; + + if (clk_get_rate(simple_dai->clk) == rate) + return 0; + + return clk_set_rate(simple_dai->clk, rate); +} + +int asoc_simple_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + simple_priv_to_props(priv, rtd->num); + unsigned int mclk, mclk_fs = 0; + int ret = 0; + + if (dai_props->mclk_fs) + mclk_fs = dai_props->mclk_fs; + + if (mclk_fs) { + mclk = params_rate(params) * mclk_fs; + + ret = asoc_simple_set_clk_rate(dai_props->codec_dai, mclk); + if (ret < 0) + return ret; + + ret = asoc_simple_set_clk_rate(dai_props->cpu_dai, mclk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) + goto err; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + goto err; + } + return 0; +err: + return ret; +} +EXPORT_SYMBOL_GPL(asoc_simple_hw_params); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bb57c482eac8..7e836473eb11 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -26,66 +26,10 @@ struct link_info { #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," -static int simple_set_clk_rate(struct asoc_simple_dai *simple_dai, - unsigned long rate) -{ - if (!simple_dai) - return 0; - - if (!simple_dai->clk) - return 0; - - if (clk_get_rate(simple_dai->clk) == rate) - return 0; - - return clk_set_rate(simple_dai->clk, rate); -} - -static int simple_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - simple_priv_to_props(priv, rtd->num); - unsigned int mclk, mclk_fs = 0; - int ret = 0; - - if (dai_props->mclk_fs) - mclk_fs = dai_props->mclk_fs; - - if (mclk_fs) { - mclk = params_rate(params) * mclk_fs; - - ret = simple_set_clk_rate(dai_props->codec_dai, mclk); - if (ret < 0) - return ret; - - ret = simple_set_clk_rate(dai_props->cpu_dai, mclk); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) - goto err; - - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret && ret != -ENOTSUPP) - goto err; - } - return 0; -err: - return ret; -} - static const struct snd_soc_ops simple_ops = { .startup = asoc_simple_startup, .shutdown = asoc_simple_shutdown, - .hw_params = simple_hw_params, + .hw_params = asoc_simple_hw_params, }; static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: share asoc_simple_dai_init()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: share asoc_simple_dai_init() 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 ad934ca8010843482d61fda46786449a9bc99e10 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:55:52 +0900 Subject: [PATCH] ASoC: simple-card-utils: share asoc_simple_dai_init() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_dai_init() between in these 2 drivers. 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 | 4 +--- sound/soc/generic/audio-graph-card.c | 23 ++--------------------- sound/soc/generic/simple-card-utils.c | 25 ++++++++++++++++++++++--- sound/soc/generic/simple-card.c | 25 +++---------------------- 4 files changed, 28 insertions(+), 49 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 89ae0841b711..204343789ae7 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -88,6 +88,7 @@ int asoc_simple_startup(struct snd_pcm_substream *substream); void asoc_simple_shutdown(struct snd_pcm_substream *substream); int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); +int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ @@ -130,9 +131,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, &(dai)->slots, \ &(dai)->slot_width); -int asoc_simple_card_init_dai(struct snd_soc_dai *dai, - struct asoc_simple_dai *simple_dai); - void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 51160adaf5bb..b1ceefc60879 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -62,25 +62,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret = 0; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -255,7 +236,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &graph_ops; - dai_link->init = graph_dai_init; + dai_link->init = asoc_simple_dai_init; return 0; } @@ -327,7 +308,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, return ret; dai_link->ops = &graph_ops; - dai_link->init = graph_dai_init; + dai_link->init = asoc_simple_dai_init; asoc_simple_card_canonicalize_platform(dai_link); asoc_simple_card_canonicalize_cpu(dai_link, diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index de23cf8bffbd..c9b53b340028 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -436,8 +436,8 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_graph_dai); -int asoc_simple_card_init_dai(struct snd_soc_dai *dai, - struct asoc_simple_dai *simple_dai) +static int asoc_simple_card_init_dai(struct snd_soc_dai *dai, + struct asoc_simple_dai *simple_dai) { int ret; @@ -467,7 +467,26 @@ int asoc_simple_card_init_dai(struct snd_soc_dai *dai, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); + +int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + int ret; + + ret = asoc_simple_card_init_dai(rtd->codec_dai, + dai_props->codec_dai); + if (ret < 0) + return ret; + + ret = asoc_simple_card_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); + if (ret < 0) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_dai_init); void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 7e836473eb11..bf4482ecfe3e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -32,25 +32,6 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; -static int simple_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - int ret; - - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); - if (ret < 0) - return ret; - - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); - if (ret < 0) - return ret; - - return 0; -} - static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -221,7 +202,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &simple_ops; - dai_link->init = simple_dai_init; + dai_link->init = asoc_simple_dai_init; return 0; } @@ -316,7 +297,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, goto dai_link_of_err; dai_link->ops = &simple_ops; - dai_link->init = simple_dai_init; + dai_link->init = asoc_simple_dai_init; asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); asoc_simple_card_canonicalize_platform(dai_link); @@ -694,7 +675,7 @@ static int simple_probe(struct platform_device *pdev) dai_link->stream_name = cinfo->name; dai_link->cpu_dai_name = cinfo->cpu_dai.name; dai_link->dai_fmt = cinfo->daifmt; - dai_link->init = simple_dai_init; + dai_link->init = asoc_simple_dai_init; memcpy(priv->dai_props->cpu_dai, &cinfo->cpu_dai, sizeof(*priv->dai_props->cpu_dai)); memcpy(priv->dai_props->codec_dai, &cinfo->codec_dai, -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup() 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 629f75440a68220a78aef9d8569831824890c47d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:56:06 +0900 Subject: [PATCH] ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph have almost same functions. This patch shares asoc_simple_be_hw_params_fixup() between in these 2 drivers. 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 | 2 ++ sound/soc/generic/audio-graph-card.c | 13 +------------ sound/soc/generic/simple-card-utils.c | 12 ++++++++++++ sound/soc/generic/simple-card.c | 13 +------------ 4 files changed, 16 insertions(+), 24 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 204343789ae7..20d41bb14ea8 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -89,6 +89,8 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream); int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); +int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); #define asoc_simple_card_parse_cpu(node, dai_link, \ list_name, cells_name, is_single_link) \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b1ceefc60879..605126fb2810 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -62,17 +62,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - return 0; -} - static void graph_parse_convert(struct device *dev, struct device_node *ep, struct asoc_simple_card_data *adata) @@ -186,7 +175,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, /* BE settings */ dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = graph_be_hw_params_fixup; + dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; dai = dai_props->codec_dai = &priv->dais[li->dais++]; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index c9b53b340028..ec4a010400a0 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -293,6 +293,18 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(asoc_simple_hw_params); +int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + + asoc_simple_card_convert_fixup(&dai_props->adata, params); + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_simple_be_hw_params_fixup); + int asoc_simple_card_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, struct device_node **dai_of_node, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bf4482ecfe3e..60a88a55c071 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -32,17 +32,6 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; -static int simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - - asoc_simple_card_convert_fixup(&dai_props->adata, params); - - return 0; -} - static void simple_parse_convert(struct device *dev, struct device_node *np, struct asoc_simple_card_data *adata) @@ -154,7 +143,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, /* BE settings */ dai_link->no_pcm = 1; - dai_link->be_hw_params_fixup = simple_be_hw_params_fixup; + dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; dai = dai_props->codec_dai = &priv->dais[li->dais++]; -- 2.20.1

1 0

[alsa-devel] [PATCH v2 00/11] ASoC: simple-card: share similar function
by Kuninori Morimoto 21 Mar '19

21 Mar '19

Hi Mark v1 patch-set had compile error at 8) patch. These are v2. Kuninori Morimoto (11): 1) ASoC: simple-card-utils: add asoc_simple_debug_info() 2) ASoC: simple_card_utils: share common priv for simple-card/audio-graph 3) ASoC: simple-card-utils: share asoc_simple_startup() 4) ASoC: simple-card-utils: share asoc_simple_shutdown() 5) ASoC: simple-card-utils: share asoc_simple_hw_param() 6) ASoC: simple-card-utils: share asoc_simple_dai_init() 7) ASoC: simple-card-utils: share asoc_simple_be_hw_params_fixup() 8) ASoC: simple-card-utils: share asoc_simple_card_init_priv() 9) ASoC: simple-card-utils: separate asoc_simple_card_parse_dai() 10) ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_() 11) ASoC: simple-card-utils: test memory allocation include/sound/simple_card_utils.h | 236 ++++++++++++++------- sound/soc/generic/audio-graph-card.c | 367 ++++++++++++++------------------ sound/soc/generic/simple-card-utils.c | 379 ++++++++++++++++++---------------- sound/soc/generic/simple-card.c | 344 ++++++++++-------------------- 4 files changed, 622 insertions(+), 704 deletions(-) -- 2.7.4 Best regards --- Kuninori Morimoto

2 12

[alsa-devel] Applied "ASoC: simple-card-utils: separate asoc_simple_card_parse_dai()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: separate asoc_simple_card_parse_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 8f7f298a333761a528e103cda3b42f3a416ad1ee Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:56:36 +0900 Subject: [PATCH] ASoC: simple-card-utils: separate asoc_simple_card_parse_dai() The difference between simple-card / audio-graph are just using OF graph style, or not. In other words, other things should be same. This means, simple-card/audio-graph common functions should be implemented at simple-card-utils, and its own functions should be implemented at each files. Current simple-card / audio-graph are using asoc_simple_card_parse_dai() which is different implementation. But, these are implemanted at simple-card-utils. It should be implemanted at each files. This patch separate these into each files. 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 | 39 ++----- sound/soc/generic/audio-graph-card.c | 114 ++++++++++++++++++-- sound/soc/generic/simple-card-utils.c | 143 -------------------------- sound/soc/generic/simple-card.c | 58 +++++++++-- 4 files changed, 164 insertions(+), 190 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index c0be02de6993..0c070c2bc119 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -99,40 +99,17 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); -#define asoc_simple_card_parse_cpu(node, dai_link, \ - list_name, cells_name, is_single_link) \ - 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) \ +#define asoc_simple_card_parse_cpu(node, dai_link, is_single_link) \ + asoc_simple_card_parse_dai(node, NULL, \ + &dai_link->cpu_of_node, \ + &dai_link->cpu_dai_name, is_single_link) +#define asoc_simple_card_parse_codec(node, dai_link) \ 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) \ + &dai_link->codec_dai_name, NULL) +#define asoc_simple_card_parse_platform(node, dai_link) \ asoc_simple_card_parse_dai(node, dai_link->platforms, \ - &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, - const char *cells_name, - int *is_single_links); - -#define asoc_simple_card_parse_graph_cpu(ep, dai_link) \ - 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->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); + &dai_link->platform_of_node, NULL, NULL) #define asoc_simple_card_of_parse_tdm(np, dai) \ snd_soc_of_parse_tdm_slot(np, &(dai)->tx_slot_mask, \ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index b9a93379098d..3370b6fc6da2 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -55,6 +55,103 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; +static int graph_get_dai_id(struct device_node *ep) +{ + struct device_node *node; + struct device_node *endpoint; + struct of_endpoint info; + int i, id; + int ret; + + /* use driver specified DAI ID if exist */ + ret = snd_soc_get_dai_id(ep); + if (ret != -ENOTSUPP) + return ret; + + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_get_property(ep, "reg", NULL)) + return info.id; + + node = of_get_parent(ep); + of_node_put(node); + if (of_get_property(node, "reg", NULL)) + return info.port; + } + node = of_graph_get_port_parent(ep); + + /* + * Non HDMI sound case, counting port/endpoint on its DT + * is enough. Let's count it. + */ + i = 0; + id = -1; + for_each_endpoint_of_node(node, endpoint) { + if (endpoint == ep) + id = i; + i++; + } + + of_node_put(node); + + if (id < 0) + return -ENODEV; + + return id; +} + +static int asoc_simple_card_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) +{ + struct device_node *node; + 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) + return 0; + + node = of_graph_get_port_parent(ep); + + /* Get dai->name */ + args.np = node; + args.args[0] = graph_get_dai_id(ep); + args.args_count = (of_graph_get_endpoint_count(node) > 1); + + ret = snd_soc_get_dai_name(&args, dai_name); + if (ret < 0) + return ret; + + *dai_of_node = node; + + if (is_single_link) + *is_single_link = of_graph_get_endpoint_count(node) == 1; + + return 0; +} + static void graph_parse_convert(struct device *dev, struct device_node *ep, struct asoc_simple_card_data *adata) @@ -128,6 +225,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, of_node_put(node); if (li->cpu) { + int is_single_links = 0; /* BE is dummy */ codecs->of_node = NULL; @@ -141,7 +239,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_graph_cpu(ep, dai_link); + ret = asoc_simple_card_parse_cpu(ep, dai_link, &is_single_links); if (ret) return ret; @@ -156,8 +254,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, return ret; /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); + asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); } else { struct snd_soc_codec_conf *cconf; @@ -176,7 +273,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_graph_codec(ep, dai_link); + ret = asoc_simple_card_parse_codec(ep, dai_link); if (ret < 0) return ret; @@ -234,7 +331,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - int ret; + int ret, single_cpu; /* Do it only CPU turn */ if (!li->cpu) @@ -258,11 +355,11 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) return ret; - ret = asoc_simple_card_parse_graph_cpu(cpu_ep, dai_link); + ret = asoc_simple_card_parse_cpu(cpu_ep, dai_link, &single_cpu); if (ret < 0) return ret; - ret = asoc_simple_card_parse_graph_codec(codec_ep, dai_link); + ret = asoc_simple_card_parse_codec(codec_ep, dai_link); if (ret < 0) return ret; @@ -293,8 +390,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, dai_link->init = asoc_simple_dai_init; asoc_simple_card_canonicalize_platform(dai_link); - asoc_simple_card_canonicalize_cpu(dai_link, - of_graph_get_endpoint_count(dai_link->cpu_of_node) == 1); + asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); return 0; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 697e820bee18..40cefbde6d9b 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -305,149 +305,6 @@ int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, } EXPORT_SYMBOL_GPL(asoc_simple_be_hw_params_fixup); -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, - const char *cells_name, - int *is_single_link) -{ - struct of_phandle_args args; - int ret; - - 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() - */ - ret = of_parse_phandle_with_args(node, list_name, cells_name, 0, &args); - if (ret) - return ret; - - /* Get dai->name */ - if (dai_name) { - ret = snd_soc_of_get_dai_name(node, dai_name); - if (ret < 0) - return ret; - } - - *dai_of_node = args.np; - - if (is_single_link) - *is_single_link = !args.args_count; - - return 0; -} -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_dai); - -static int asoc_simple_card_get_dai_id(struct device_node *ep) -{ - struct device_node *node; - struct device_node *endpoint; - struct of_endpoint info; - int i, id; - int ret; - - /* use driver specified DAI ID if exist */ - ret = snd_soc_get_dai_id(ep); - if (ret != -ENOTSUPP) - return ret; - - /* use endpoint/port reg if exist */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret == 0) { - /* - * Because it will count port/endpoint if it doesn't have "reg". - * But, we can't judge whether it has "no reg", or "reg = <0>" - * only of_graph_parse_endpoint(). - * We need to check "reg" property - */ - if (of_get_property(ep, "reg", NULL)) - return info.id; - - node = of_get_parent(ep); - of_node_put(node); - if (of_get_property(node, "reg", NULL)) - return info.port; - } - node = of_graph_get_port_parent(ep); - - /* - * Non HDMI sound case, counting port/endpoint on its DT - * is enough. Let's count it. - */ - i = 0; - id = -1; - for_each_endpoint_of_node(node, endpoint) { - if (endpoint == ep) - id = i; - i++; - } - - of_node_put(node); - - if (id < 0) - return -ENODEV; - - return id; -} - -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) -{ - struct device_node *node; - 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) - return 0; - - node = of_graph_get_port_parent(ep); - - /* Get dai->name */ - args.np = node; - args.args[0] = asoc_simple_card_get_dai_id(ep); - args.args_count = (of_graph_get_endpoint_count(node) > 1); - - ret = snd_soc_get_dai_name(&args, dai_name); - if (ret < 0) - return ret; - - *dai_of_node = node; - - return 0; -} -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_graph_dai); - static int asoc_simple_card_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai) { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4e3e6b34593c..d8560fb1f5de 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,6 +25,52 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; +static 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, + int *is_single_link) +{ + struct of_phandle_args args; + int ret; + + 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() + */ + ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args); + if (ret) + return ret; + + /* Get dai->name */ + if (dai_name) { + ret = snd_soc_of_get_dai_name(node, dai_name); + if (ret < 0) + return ret; + } + + *dai_of_node = args.np; + + if (is_single_link) + *is_single_link = !args.args_count; + + return 0; +} + static void simple_parse_convert(struct device *dev, struct device_node *np, struct asoc_simple_card_data *adata) @@ -110,8 +156,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_cpu(np, dai_link, DAI, CELL, - &is_single_links); + ret = asoc_simple_card_parse_cpu(np, dai_link, &is_single_links); if (ret) return ret; @@ -144,7 +189,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_codec(np, dai_link, DAI, CELL); + ret = asoc_simple_card_parse_codec(np, dai_link); if (ret < 0) return ret; @@ -242,16 +287,15 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); - ret = asoc_simple_card_parse_cpu(cpu, dai_link, - DAI, CELL, &single_cpu); + ret = asoc_simple_card_parse_cpu(cpu, dai_link, &single_cpu); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_codec(codec, dai_link, DAI, CELL); + ret = asoc_simple_card_parse_codec(codec, dai_link); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_platform(plat, dai_link, DAI, CELL); + ret = asoc_simple_card_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_()" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_() 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 ad11e59f52d6fc75037ac3cb66dc711b83c1bbf8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:56:50 +0900 Subject: [PATCH] ASoC: simple-card-utils: rename asoc_simple_card_xxx() to asoc_simple_() Current simple-card-utils is using asoc_simple_card_xxx() for each function naming, but it is very verbose. Thus it is easy to be over 80 char. This patch renames it to asoc_simple_xxx(). 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 | 90 +++++++++---------- sound/soc/generic/audio-graph-card.c | 98 ++++++++++----------- sound/soc/generic/simple-card-utils.c | 122 +++++++++++++------------- sound/soc/generic/simple-card.c | 98 ++++++++++----------- 4 files changed, 204 insertions(+), 204 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0c070c2bc119..67dc3ee6ed01 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -10,10 +10,10 @@ #include <sound/soc.h> -#define asoc_simple_card_init_hp(card, sjack, prefix) \ - asoc_simple_card_init_jack(card, sjack, 1, prefix) -#define asoc_simple_card_init_mic(card, sjack, prefix) \ - asoc_simple_card_init_jack(card, sjack, 0, prefix) +#define asoc_simple_init_hp(card, sjack, prefix) \ + asoc_simple_init_jack(card, sjack, 1, prefix) +#define asoc_simple_init_mic(card, sjack, prefix) \ + asoc_simple_init_jack(card, sjack, 0, prefix) struct asoc_simple_dai { const char *name; @@ -26,7 +26,7 @@ struct asoc_simple_dai { struct clk *clk; }; -struct asoc_simple_card_data { +struct asoc_simple_data { u32 convert_rate; u32 convert_channels; }; @@ -44,7 +44,7 @@ struct asoc_simple_priv { struct asoc_simple_dai *codec_dai; struct snd_soc_dai_link_component codecs; /* single codec */ struct snd_soc_dai_link_component platforms; - struct asoc_simple_card_data adata; + struct asoc_simple_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; } *dai_props; @@ -67,30 +67,30 @@ struct link_info { int cpu; /* turn for CPU / Codec */ }; -int asoc_simple_card_parse_daifmt(struct device *dev, - struct device_node *node, - struct device_node *codec, - char *prefix, - unsigned int *retfmt); +int asoc_simple_parse_daifmt(struct device *dev, + struct device_node *node, + struct device_node *codec, + char *prefix, + unsigned int *retfmt); __printf(3, 4) -int asoc_simple_card_set_dailink_name(struct device *dev, - struct snd_soc_dai_link *dai_link, - const char *fmt, ...); -int asoc_simple_card_parse_card_name(struct snd_soc_card *card, - char *prefix); - -#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, \ +int asoc_simple_set_dailink_name(struct device *dev, + struct snd_soc_dai_link *dai_link, + const char *fmt, ...); +int asoc_simple_parse_card_name(struct snd_soc_card *card, + char *prefix); + +#define asoc_simple_parse_clk_cpu(dev, node, dai_link, simple_dai) \ + asoc_simple_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai, \ 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,\ +#define asoc_simple_parse_clk_codec(dev, node, dai_link, simple_dai) \ + asoc_simple_parse_clk(dev, node, dai_link->codec_of_node, simple_dai,\ 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 *dai_name, - struct snd_soc_dai_link_component *dlc); +int asoc_simple_parse_clk(struct device *dev, + struct device_node *node, + struct device_node *dai_of_node, + struct asoc_simple_dai *simple_dai, + const char *dai_name, + struct snd_soc_dai_link_component *dlc); int asoc_simple_startup(struct snd_pcm_substream *substream); void asoc_simple_shutdown(struct snd_pcm_substream *substream); int asoc_simple_hw_params(struct snd_pcm_substream *substream, @@ -99,45 +99,45 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd); int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); -#define asoc_simple_card_parse_cpu(node, dai_link, is_single_link) \ - asoc_simple_card_parse_dai(node, NULL, \ +#define asoc_simple_parse_cpu(node, dai_link, is_single_link) \ + asoc_simple_parse_dai(node, NULL, \ &dai_link->cpu_of_node, \ &dai_link->cpu_dai_name, is_single_link) -#define asoc_simple_card_parse_codec(node, dai_link) \ - asoc_simple_card_parse_dai(node, dai_link->codecs, \ +#define asoc_simple_parse_codec(node, dai_link) \ + asoc_simple_parse_dai(node, dai_link->codecs, \ &dai_link->codec_of_node, \ &dai_link->codec_dai_name, NULL) -#define asoc_simple_card_parse_platform(node, dai_link) \ - asoc_simple_card_parse_dai(node, dai_link->platforms, \ +#define asoc_simple_parse_platform(node, dai_link) \ + asoc_simple_parse_dai(node, dai_link->platforms, \ &dai_link->platform_of_node, NULL, NULL) -#define asoc_simple_card_of_parse_tdm(np, dai) \ +#define asoc_simple_parse_tdm(np, dai) \ snd_soc_of_parse_tdm_slot(np, &(dai)->tx_slot_mask, \ &(dai)->rx_slot_mask, \ &(dai)->slots, \ &(dai)->slot_width); -void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link); -void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, +void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link); +void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int is_single_links); -int asoc_simple_card_clean_reference(struct snd_soc_card *card); +int asoc_simple_clean_reference(struct snd_soc_card *card); -void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, +void asoc_simple_convert_fixup(struct asoc_simple_data *data, struct snd_pcm_hw_params *params); -void asoc_simple_card_parse_convert(struct device *dev, - struct device_node *np, char *prefix, - struct asoc_simple_card_data *data); +void asoc_simple_parse_convert(struct device *dev, + struct device_node *np, char *prefix, + struct asoc_simple_data *data); -int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, +int asoc_simple_parse_routing(struct snd_soc_card *card, char *prefix); -int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, +int asoc_simple_parse_widgets(struct snd_soc_card *card, char *prefix); -int asoc_simple_card_init_jack(struct snd_soc_card *card, +int asoc_simple_init_jack(struct snd_soc_card *card, struct asoc_simple_jack *sjack, int is_hp, char *prefix); -int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, +int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct link_info *li); #ifdef DEBUG diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3370b6fc6da2..8114b5ea9319 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -107,11 +107,11 @@ static int graph_get_dai_id(struct device_node *ep) return id; } -static int asoc_simple_card_parse_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - struct device_node **dai_of_node, - const char **dai_name, - int *is_single_link) +static int asoc_simple_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) { struct device_node *node; struct of_phandle_args args; @@ -154,18 +154,18 @@ static int asoc_simple_card_parse_dai(struct device_node *ep, static void graph_parse_convert(struct device *dev, struct device_node *ep, - struct asoc_simple_card_data *adata) + struct asoc_simple_data *adata) { struct device_node *top = dev->of_node; struct device_node *port = of_get_parent(ep); struct device_node *ports = of_get_parent(port); struct device_node *node = of_graph_get_port_parent(ep); - asoc_simple_card_parse_convert(dev, top, NULL, adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, adata); - asoc_simple_card_parse_convert(dev, ports, NULL, adata); - asoc_simple_card_parse_convert(dev, port, NULL, adata); - asoc_simple_card_parse_convert(dev, ep, NULL, adata); + asoc_simple_parse_convert(dev, top, NULL, adata); + asoc_simple_parse_convert(dev, node, PREFIX, adata); + asoc_simple_parse_convert(dev, ports, NULL, adata); + asoc_simple_parse_convert(dev, port, NULL, adata); + asoc_simple_parse_convert(dev, ep, NULL, adata); of_node_put(port); of_node_put(ports); @@ -239,22 +239,22 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_cpu(ep, dai_link, &is_single_links); + ret = asoc_simple_parse_cpu(ep, dai_link, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, ep, dai_link, dai); + ret = asoc_simple_parse_clk_cpu(dev, ep, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); if (ret < 0) return ret; /* card->num_links includes Codec */ - asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); + asoc_simple_canonicalize_cpu(dai_link, is_single_links); } else { struct snd_soc_codec_conf *cconf; @@ -273,17 +273,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_codec(ep, dai_link); + ret = asoc_simple_parse_codec(ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, ep, dai_link, dai); + ret = asoc_simple_parse_clk_codec(dev, ep, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); if (ret < 0) return ret; @@ -301,14 +301,14 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, graph_parse_convert(dev, ep, &dai_props->adata); graph_parse_mclk_fs(top, ep, dai_props); - asoc_simple_card_canonicalize_platform(dai_link); + asoc_simple_canonicalize_platform(dai_link); - ret = asoc_simple_card_of_parse_tdm(ep, dai); + ret = asoc_simple_parse_tdm(ep, dai); if (ret) return ret; - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); if (ret < 0) return ret; @@ -350,47 +350,47 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, graph_parse_mclk_fs(top, cpu_ep, dai_props); graph_parse_mclk_fs(top, codec_ep, dai_props); - ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep, - NULL, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep, + NULL, &dai_link->dai_fmt); if (ret < 0) return ret; - ret = asoc_simple_card_parse_cpu(cpu_ep, dai_link, &single_cpu); + ret = asoc_simple_parse_cpu(cpu_ep, dai_link, &single_cpu); if (ret < 0) return ret; - ret = asoc_simple_card_parse_codec(codec_ep, dai_link); + ret = asoc_simple_parse_codec(codec_ep, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_tdm(cpu_ep, cpu_dai); + ret = asoc_simple_parse_tdm(cpu_ep, cpu_dai); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_tdm(codec_ep, codec_dai); + ret = asoc_simple_parse_tdm(codec_ep, codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, cpu_ep, dai_link, cpu_dai); + ret = asoc_simple_parse_clk_cpu(dev, cpu_ep, dai_link, cpu_dai); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, codec_ep, dai_link, codec_dai); + ret = asoc_simple_parse_clk_codec(dev, codec_ep, dai_link, codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "%s-%s", - dai_link->cpu_dai_name, - dai_link->codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "%s-%s", + dai_link->cpu_dai_name, + dai_link->codecs->dai_name); if (ret < 0) return ret; dai_link->ops = &graph_ops; dai_link->init = asoc_simple_dai_init; - asoc_simple_card_canonicalize_platform(dai_link); - asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_canonicalize_platform(dai_link); return 0; } @@ -414,7 +414,7 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old = NULL; - struct asoc_simple_card_data adata; + struct asoc_simple_data adata; int rc, ret; /* loop for all listed CPU port */ @@ -469,11 +469,11 @@ static int graph_parse_of(struct asoc_simple_priv *priv) struct link_info li; int ret; - ret = asoc_simple_card_of_parse_widgets(card, NULL); + ret = asoc_simple_parse_widgets(card, NULL); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_routing(card, NULL); + ret = asoc_simple_parse_routing(card, NULL); if (ret < 0) return ret; @@ -498,7 +498,7 @@ static int graph_parse_of(struct asoc_simple_priv *priv) return ret; } - return asoc_simple_card_parse_card_name(card, NULL); + return asoc_simple_parse_card_name(card, NULL); } static int graph_count_noml(struct asoc_simple_priv *priv, @@ -601,11 +601,11 @@ static int graph_card_probe(struct snd_soc_card *card) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; - ret = asoc_simple_card_init_hp(card, &priv->hp_jack, NULL); + ret = asoc_simple_init_hp(card, &priv->hp_jack, NULL); if (ret < 0) return ret; - ret = asoc_simple_card_init_mic(card, &priv->mic_jack, NULL); + ret = asoc_simple_init_mic(card, &priv->mic_jack, NULL); if (ret < 0) return ret; @@ -637,7 +637,7 @@ static int graph_probe(struct platform_device *pdev) if (!li.link || !li.dais) return -EINVAL; - ret = asoc_simple_card_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, &li); if (ret < 0) return ret; @@ -665,7 +665,7 @@ static int graph_probe(struct platform_device *pdev) return 0; err: - asoc_simple_card_clean_reference(card); + asoc_simple_clean_reference(card); return ret; } @@ -674,7 +674,7 @@ static int graph_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_card_clean_reference(card); + return asoc_simple_clean_reference(card); } static const struct of_device_id graph_of_match[] = { diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 40cefbde6d9b..a794f767ff7f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -14,8 +14,8 @@ #include <sound/jack.h> #include <sound/simple_card_utils.h> -void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, - struct snd_pcm_hw_params *params) +void asoc_simple_convert_fixup(struct asoc_simple_data *data, + struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -30,12 +30,12 @@ void asoc_simple_card_convert_fixup(struct asoc_simple_card_data *data, channels->min = channels->max = data->convert_channels; } -EXPORT_SYMBOL_GPL(asoc_simple_card_convert_fixup); +EXPORT_SYMBOL_GPL(asoc_simple_convert_fixup); -void asoc_simple_card_parse_convert(struct device *dev, - struct device_node *np, - char *prefix, - struct asoc_simple_card_data *data) +void asoc_simple_parse_convert(struct device *dev, + struct device_node *np, + char *prefix, + struct asoc_simple_data *data) { char prop[128]; @@ -50,13 +50,13 @@ void asoc_simple_card_parse_convert(struct device *dev, snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); of_property_read_u32(np, prop, &data->convert_channels); } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_convert); +EXPORT_SYMBOL_GPL(asoc_simple_parse_convert); -int asoc_simple_card_parse_daifmt(struct device *dev, - struct device_node *node, - struct device_node *codec, - char *prefix, - unsigned int *retfmt) +int asoc_simple_parse_daifmt(struct device *dev, + struct device_node *node, + struct device_node *codec, + char *prefix, + unsigned int *retfmt) { struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; @@ -92,11 +92,11 @@ int asoc_simple_card_parse_daifmt(struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_daifmt); +EXPORT_SYMBOL_GPL(asoc_simple_parse_daifmt); -int asoc_simple_card_set_dailink_name(struct device *dev, - struct snd_soc_dai_link *dai_link, - const char *fmt, ...) +int asoc_simple_set_dailink_name(struct device *dev, + struct snd_soc_dai_link *dai_link, + const char *fmt, ...) { va_list ap; char *name = NULL; @@ -115,10 +115,10 @@ int asoc_simple_card_set_dailink_name(struct device *dev, return ret; } -EXPORT_SYMBOL_GPL(asoc_simple_card_set_dailink_name); +EXPORT_SYMBOL_GPL(asoc_simple_set_dailink_name); -int asoc_simple_card_parse_card_name(struct snd_soc_card *card, - char *prefix) +int asoc_simple_parse_card_name(struct snd_soc_card *card, + char *prefix) { int ret; @@ -141,9 +141,9 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); +EXPORT_SYMBOL_GPL(asoc_simple_parse_card_name); -static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) +static int asoc_simple_clk_enable(struct asoc_simple_dai *dai) { if (dai) return clk_prepare_enable(dai->clk); @@ -151,18 +151,18 @@ static int asoc_simple_card_clk_enable(struct asoc_simple_dai *dai) return 0; } -static void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai) +static void asoc_simple_clk_disable(struct asoc_simple_dai *dai) { if (dai) clk_disable_unprepare(dai->clk); } -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 *dai_name, - struct snd_soc_dai_link_component *dlc) +int asoc_simple_parse_clk(struct device *dev, + struct device_node *node, + struct device_node *dai_of_node, + struct asoc_simple_dai *simple_dai, + const char *dai_name, + struct snd_soc_dai_link_component *dlc) { struct clk *clk; u32 val; @@ -202,7 +202,7 @@ int asoc_simple_card_parse_clk(struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_parse_clk); +EXPORT_SYMBOL_GPL(asoc_simple_parse_clk); int asoc_simple_startup(struct snd_pcm_substream *substream) { @@ -211,13 +211,13 @@ int asoc_simple_startup(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_clk_enable(dai_props->cpu_dai); + ret = asoc_simple_clk_enable(dai_props->cpu_dai); if (ret) return ret; - ret = asoc_simple_card_clk_enable(dai_props->codec_dai); + ret = asoc_simple_clk_enable(dai_props->codec_dai); if (ret) - asoc_simple_card_clk_disable(dai_props->cpu_dai); + asoc_simple_clk_disable(dai_props->cpu_dai); return ret; } @@ -230,9 +230,9 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_clk_disable(dai_props->cpu_dai); + asoc_simple_clk_disable(dai_props->cpu_dai); - asoc_simple_card_clk_disable(dai_props->codec_dai); + asoc_simple_clk_disable(dai_props->codec_dai); } EXPORT_SYMBOL_GPL(asoc_simple_shutdown); @@ -299,13 +299,13 @@ int asoc_simple_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); - asoc_simple_card_convert_fixup(&dai_props->adata, params); + asoc_simple_convert_fixup(&dai_props->adata, params); return 0; } EXPORT_SYMBOL_GPL(asoc_simple_be_hw_params_fixup); -static int asoc_simple_card_init_dai(struct snd_soc_dai *dai, +static int asoc_simple_init_dai(struct snd_soc_dai *dai, struct asoc_simple_dai *simple_dai) { int ret; @@ -343,13 +343,13 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); int ret; - ret = asoc_simple_card_init_dai(rtd->codec_dai, - dai_props->codec_dai); + ret = asoc_simple_init_dai(rtd->codec_dai, + dai_props->codec_dai); if (ret < 0) return ret; - ret = asoc_simple_card_init_dai(rtd->cpu_dai, - dai_props->cpu_dai); + ret = asoc_simple_init_dai(rtd->cpu_dai, + dai_props->cpu_dai); if (ret < 0) return ret; @@ -357,16 +357,16 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); -void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link) +void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ if (!dai_link->platforms->of_node) dai_link->platforms->of_node = dai_link->cpu_of_node; } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_platform); +EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform); -void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, - int is_single_links) +void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link *dai_link, + int is_single_links) { /* * In soc_bind_dai_link() will check cpu name after @@ -380,9 +380,9 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, if (is_single_links) dai_link->cpu_dai_name = NULL; } -EXPORT_SYMBOL_GPL(asoc_simple_card_canonicalize_cpu); +EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_cpu); -int asoc_simple_card_clean_reference(struct snd_soc_card *card) +int asoc_simple_clean_reference(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; int i; @@ -393,10 +393,10 @@ int asoc_simple_card_clean_reference(struct snd_soc_card *card) } return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_clean_reference); +EXPORT_SYMBOL_GPL(asoc_simple_clean_reference); -int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, - char *prefix) +int asoc_simple_parse_routing(struct snd_soc_card *card, + char *prefix) { struct device_node *node = card->dev->of_node; char prop[128]; @@ -411,10 +411,10 @@ int asoc_simple_card_of_parse_routing(struct snd_soc_card *card, return snd_soc_of_parse_audio_routing(card, prop); } -EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_routing); +EXPORT_SYMBOL_GPL(asoc_simple_parse_routing); -int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, - char *prefix) +int asoc_simple_parse_widgets(struct snd_soc_card *card, + char *prefix) { struct device_node *node = card->dev->of_node; char prop[128]; @@ -430,11 +430,11 @@ int asoc_simple_card_of_parse_widgets(struct snd_soc_card *card, /* no widgets is not error */ return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_of_parse_widgets); +EXPORT_SYMBOL_GPL(asoc_simple_parse_widgets); -int asoc_simple_card_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, - int is_hp, char *prefix) +int asoc_simple_init_jack(struct snd_soc_card *card, + struct asoc_simple_jack *sjack, + int is_hp, char *prefix) { struct device *dev = card->dev; enum of_gpio_flags flags; @@ -485,10 +485,10 @@ int asoc_simple_card_init_jack(struct snd_soc_card *card, return 0; } -EXPORT_SYMBOL_GPL(asoc_simple_card_init_jack); +EXPORT_SYMBOL_GPL(asoc_simple_init_jack); -int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, - struct link_info *li) +int asoc_simple_init_priv(struct asoc_simple_priv *priv, + struct link_info *li) { struct snd_soc_card *card = simple_priv_to_card(priv); struct device *dev = simple_priv_to_dev(priv); @@ -513,7 +513,7 @@ int asoc_simple_card_init_priv(struct asoc_simple_priv *priv, * * "platform" might be removed * see - * simple-card-utils.c :: asoc_simple_card_canonicalize_platform() + * simple-card-utils.c :: asoc_simple_canonicalize_platform() */ for (i = 0; i < li->link; i++) { dai_link[i].codecs = &dai_props[i].codecs; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d8560fb1f5de..dd9ac60d61f1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,11 +25,11 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; -static 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, - int *is_single_link) +static int asoc_simple_parse_dai(struct device_node *node, + struct snd_soc_dai_link_component *dlc, + struct device_node **dai_of_node, + const char **dai_name, + int *is_single_link) { struct of_phandle_args args; int ret; @@ -73,15 +73,15 @@ static int asoc_simple_card_parse_dai(struct device_node *node, static void simple_parse_convert(struct device *dev, struct device_node *np, - struct asoc_simple_card_data *adata) + struct asoc_simple_data *adata) { struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); - asoc_simple_card_parse_convert(dev, top, PREFIX, adata); - asoc_simple_card_parse_convert(dev, node, PREFIX, adata); - asoc_simple_card_parse_convert(dev, node, NULL, adata); - asoc_simple_card_parse_convert(dev, np, NULL, adata); + asoc_simple_parse_convert(dev, top, PREFIX, adata); + asoc_simple_parse_convert(dev, node, PREFIX, adata); + asoc_simple_parse_convert(dev, node, NULL, adata); + asoc_simple_parse_convert(dev, np, NULL, adata); of_node_put(node); } @@ -156,21 +156,21 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai = dai_props->cpu_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_cpu(np, dai_link, &is_single_links); + ret = asoc_simple_parse_cpu(np, dai_link, &is_single_links); if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai); + ret = asoc_simple_parse_clk_cpu(dev, np, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "fe.%s", - dai_link->cpu_dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "fe.%s", + dai_link->cpu_dai_name); if (ret < 0) return ret; - asoc_simple_card_canonicalize_cpu(dai_link, is_single_links); + asoc_simple_canonicalize_cpu(dai_link, is_single_links); } else { struct snd_soc_codec_conf *cconf; @@ -189,17 +189,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, cconf = dai_props->codec_conf = &priv->codec_conf[li->conf++]; - ret = asoc_simple_card_parse_codec(np, dai_link); + ret = asoc_simple_parse_codec(np, dai_link); if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai); + ret = asoc_simple_parse_clk_codec(dev, np, dai_link, dai); if (ret < 0) return ret; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "be.%s", - codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); if (ret < 0) return ret; @@ -215,14 +215,14 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, simple_parse_convert(dev, np, &dai_props->adata); simple_parse_mclk_fs(top, np, codec, dai_props, prefix); - asoc_simple_card_canonicalize_platform(dai_link); + asoc_simple_canonicalize_platform(dai_link); - ret = asoc_simple_card_of_parse_tdm(np, dai); + ret = asoc_simple_parse_tdm(np, dai); if (ret) return ret; - ret = asoc_simple_card_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) return ret; @@ -280,53 +280,53 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; - ret = asoc_simple_card_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) goto dai_link_of_err; simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); - ret = asoc_simple_card_parse_cpu(cpu, dai_link, &single_cpu); + ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_codec(codec, dai_link); + ret = asoc_simple_parse_codec(codec, dai_link); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_platform(plat, dai_link); + ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_of_parse_tdm(cpu, cpu_dai); + ret = asoc_simple_parse_tdm(cpu, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_of_parse_tdm(codec, codec_dai); + ret = asoc_simple_parse_tdm(codec, codec_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_clk_cpu(dev, cpu, dai_link, cpu_dai); + ret = asoc_simple_parse_clk_cpu(dev, cpu, dai_link, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_clk_codec(dev, codec, dai_link, codec_dai); + ret = asoc_simple_parse_clk_codec(dev, codec, dai_link, codec_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_set_dailink_name(dev, dai_link, - "%s-%s", - dai_link->cpu_dai_name, - dai_link->codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "%s-%s", + dai_link->cpu_dai_name, + dai_link->codecs->dai_name); if (ret < 0) goto dai_link_of_err; dai_link->ops = &simple_ops; dai_link->init = asoc_simple_dai_init; - asoc_simple_card_canonicalize_cpu(dai_link, single_cpu); - asoc_simple_card_canonicalize_platform(dai_link); + asoc_simple_canonicalize_cpu(dai_link, single_cpu); + asoc_simple_canonicalize_platform(dai_link); dai_link_of_err: of_node_put(plat); @@ -361,7 +361,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, /* loop for all dai-link */ do { - struct asoc_simple_card_data adata; + struct asoc_simple_data adata; struct device_node *codec; struct device_node *np; int num = of_get_child_count(node); @@ -451,11 +451,11 @@ static int simple_parse_of(struct asoc_simple_priv *priv) if (!top) return -EINVAL; - ret = asoc_simple_card_of_parse_widgets(card, PREFIX); + ret = asoc_simple_parse_widgets(card, PREFIX); if (ret < 0) return ret; - ret = asoc_simple_card_of_parse_routing(card, PREFIX); + ret = asoc_simple_parse_routing(card, PREFIX); if (ret < 0) return ret; @@ -481,7 +481,7 @@ static int simple_parse_of(struct asoc_simple_priv *priv) return ret; } - ret = asoc_simple_card_parse_card_name(card, PREFIX); + ret = asoc_simple_parse_card_name(card, PREFIX); if (ret < 0) return ret; @@ -587,11 +587,11 @@ static int simple_soc_probe(struct snd_soc_card *card) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; - ret = asoc_simple_card_init_hp(card, &priv->hp_jack, PREFIX); + ret = asoc_simple_init_hp(card, &priv->hp_jack, PREFIX); if (ret < 0) return ret; - ret = asoc_simple_card_init_mic(card, &priv->mic_jack, PREFIX); + ret = asoc_simple_init_mic(card, &priv->mic_jack, PREFIX); if (ret < 0) return ret; @@ -622,7 +622,7 @@ static int simple_probe(struct platform_device *pdev) if (!li.link || !li.dais) return -EINVAL; - ret = asoc_simple_card_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, &li); if (ret < 0) return ret; @@ -691,7 +691,7 @@ static int simple_probe(struct platform_device *pdev) return 0; err: - asoc_simple_card_clean_reference(card); + asoc_simple_clean_reference(card); return ret; } @@ -700,7 +700,7 @@ static int simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - return asoc_simple_card_clean_reference(card); + return asoc_simple_clean_reference(card); } static const struct of_device_id simple_of_match[] = { -- 2.20.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: mchp-i2s-mcc: dt-bindings: add DT bindings for I2S Multi-Channel Controller
by Codrin.Ciubotariu＠microchip.com 21 Mar '19

21 Mar '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> This patch adds DT bindings for the new Microchip I2S Multi-Channel controller embedded inside sam9x60 SoCs. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- .../bindings/sound/mchp-i2s-mcc.txt | 43 +++++++++++++++++++ 1 file changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt diff --git a/Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt b/Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt new file mode 100644 index 000000000000..91ec83a6faed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt @@ -0,0 +1,43 @@ +* Microchip I2S Multi-Channel Controller + +Required properties: +- compatible: Should be "microchip,sam9x60-i2smcc". +- reg: Should be the physical base address of the controller and the + length of memory mapped region. +- interrupts: Should contain the interrupt for the controller. +- dmas: Should be one per channel name listed in the dma-names property, + as described in atmel-dma.txt and dma.txt files. +- dma-names: Identifier string for each DMA request line in the dmas property. + Two dmas have to be defined, "tx" and "rx". +- clocks: Must contain an entry for each entry in clock-names. + Please refer to clock-bindings.txt. +- clock-names: Should be one of each entry matching the clocks phandles list: + - "pclk" (peripheral clock) Required. + - "gclk" (generated clock) Optional (1). + +Optional properties: +- pinctrl-0: Should specify pin control groups used for this controller. +- princtrl-names: Should contain only one value - "default". + + +(1) : Only the peripheral clock is required. The generated clock is optional + and should be set mostly when Master Mode is required. + +Example: + + i2s@f001c000 { + compatible = "microchip,sam9x60-i2smcc"; + reg = <0xf001c000 0x100>; + interrupts = <34 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(36))>, + <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) | + AT91_XDMAC_DT_PERID(37))>; + dma-names = "tx", "rx"; + clocks = <&i2s_clk>, <&i2s_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2s_default>; + }; -- 2.17.1

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[alsa-devel] [PATCH v2 1/2] ASoC: lochnagar: Add device tree binding document
by Charles Keepax 21 Mar '19

21 Mar '19

Lochnagar is an evaluation and development board for Cirrus Logic Smart CODEC and Amp devices. It allows the connection of most Cirrus Logic devices on mini-cards, as well as allowing connection of various application processor systems to provide a full evaluation platform. This driver supports the board controller chip on the Lochnagar board. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Made clock name a bit more generic Thanks, Charles .../devicetree/bindings/sound/cirrus,lochnagar.txt | 39 ++++++++++++++++++++++ 1 file changed, 39 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt new file mode 100644 index 0000000000000..41ae2699f07a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt @@ -0,0 +1,39 @@ +Cirrus Logic Lochnagar Audio Development Board + +Lochnagar is an evaluation and development board for Cirrus Logic +Smart CODEC and Amp devices. It allows the connection of most Cirrus +Logic devices on mini-cards, as well as allowing connection of +various application processor systems to provide a full evaluation +platform. Audio system topology, clocking and power can all be +controlled through the Lochnagar, allowing the device under test +to be used in a variety of possible use cases. + +This binding document describes the binding for the audio portion +of the driver. + +This binding must be part of the Lochnagar MFD binding: + [4] ../mfd/cirrus,lochnagar.txt + +Required properties: + + - compatible : One of the following strings: + "cirrus,lochnagar2-soundcard" + + - #sound-dai-cells : Must be set to 1. + + - clocks : Contains an entry for each entry in clock-names. + - clock-names : Must include the following clocks: + "mclk" Master clock source for the sound card, should normally + be set to LOCHNAGAR_SOUNDCARD_MCLK provided by the Lochnagar + clock driver. + +Example: + +lochnagar-sc { + compatible = "cirrus,lochnagar2-soundcard"; + + #sound-dai-cells = <1>; + + clocks = <&lochnagar_clk LOCHNAGAR_SOUNDCARD_MCLK>; + clock-names = "mclk"; +}; -- 2.11.0

2 3

[alsa-devel] Applied "ASoC: simple-card-utils: test memory allocation" to the asoc tree
by Mark Brown 21 Mar '19

21 Mar '19

The patch ASoC: simple-card-utils: test memory allocation 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 008fe4e5382293e2da96a4dc75b95782a5f95be8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 20 Mar 2019 13:57:02 +0900 Subject: [PATCH] ASoC: simple-card-utils: test memory allocation li->conf will be 0 if it was not DPCM case. Then, 1) we shouldn't call devm_kcalloc() with size 0, 2) we need NULL pointer check if li->conf was not 0. This patch fixed above issues. Special thanks to Pierre-Louis Bossart Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> 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 | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index a794f767ff7f..4ed68348f939 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -495,16 +495,21 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct snd_soc_dai_link *dai_link; struct simple_dai_props *dai_props; struct asoc_simple_dai *dais; - struct snd_soc_codec_conf *cconf; + struct snd_soc_codec_conf *cconf = NULL; int i; dai_props = devm_kcalloc(dev, li->link, sizeof(*dai_props), GFP_KERNEL); dai_link = devm_kcalloc(dev, li->link, sizeof(*dai_link), GFP_KERNEL); dais = devm_kcalloc(dev, li->dais, sizeof(*dais), GFP_KERNEL); - cconf = devm_kcalloc(dev, li->conf, sizeof(*cconf), GFP_KERNEL); if (!dai_props || !dai_link || !dais) return -ENOMEM; + if (li->conf) { + cconf = devm_kcalloc(dev, li->conf, sizeof(*cconf), GFP_KERNEL); + if (!cconf) + 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. -- 2.20.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: Mediatek: Add mt8183-mt6358 machine driver
by Shunli Wang 21 Mar '19

21 Mar '19

This patch adds machine driver which uses MT8183 as SoC and MT6358 as external codec. The patch is based on broonie tree "for-next" branch Shunli Wang (2): ASoC: mediatek: mt8183: add mt8183-mt6358 machine driver ASoC: mediatek: add document for mt8183-mt6358 .../devicetree/bindings/sound/mt8183-mt6358.txt | 12 ++ sound/soc/mediatek/Kconfig | 10 + sound/soc/mediatek/mt8183/mt8183-mt6358.c | 201 +++++++++++++++++++++ 3 files changed, 223 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8183-mt6358.txt create mode 100644 sound/soc/mediatek/mt8183/mt8183-mt6358.c -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: hda/realtek - Add support for Acer Aspire E5-523G/ES1-432 headset mic
by Jian-Hong Pan 21 Mar '19

21 Mar '19

From: Chris Chiu <chiu(a)endlessm.com> The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu <chiu(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 191830d4fa40..89a96442eb07 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6712,6 +6712,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headset MIC of Acer Aspire Z24-890 with ALC286
by Jian-Hong Pan 21 Mar '19

21 Mar '19

The Acer Aspire Z24-890 cannot detect the headset MIC until ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 191830d4fa40..916bb5a3f324 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6715,6 +6715,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: seq: oss: Fix Spectre v1 vulnerability
by Gustavo A. R. Silva 21 Mar '19

21 Mar '19

dev is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap) Fix this by sanitizing dev before using it to index dp->synths. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/seq/oss/seq_oss_synth.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 278ebb993122..c93945917235 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -617,13 +617,14 @@ int snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_info *inf) { struct seq_oss_synth *rec; + struct seq_oss_synthinfo *info = get_synthinfo_nospec(dp, dev); - if (dev < 0 || dev >= dp->max_synthdev) + if (!info) return -ENXIO; - if (dp->synths[dev].is_midi) { + if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf); + snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; -- 2.21.0

2 1

[alsa-devel] [PATCH] ALSA: rawmidi: Fix potential Spectre v1 vulnerability
by Gustavo A. R. Silva 21 Mar '19

21 Mar '19

info->stream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap) Fix this by sanitizing info->stream before using it to index rmidi->streams. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/rawmidi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index ee601d7f0926..c0690d1ecd55 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -30,6 +30,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mm.h> +#include <linux/nospec.h> #include <sound/rawmidi.h> #include <sound/info.h> #include <sound/control.h> @@ -601,6 +602,7 @@ static int __snd_rawmidi_info_select(struct snd_card *card, return -ENXIO; if (info->stream < 0 || info->stream > 1) return -EINVAL; + info->stream = array_index_nospec(info->stream, 2); pstr = &rmidi->streams[info->stream]; if (pstr->substream_count == 0) return -ENOENT; -- 2.21.0

2 1

[alsa-devel] [PATCH v2 -next] ASoC: wm_adsp: Make some variables static
by Yue Haibing 21 Mar '19

21 Mar '19

From: YueHaibing <yuehaibing(a)huawei.com> Fix sparse warnings: sound/soc/codecs/wm_adsp.c:309:20: warning: symbol 'wm_adsp1_ops' was not declared. Should it be static? sound/soc/codecs/wm_adsp.c:310:20: warning: symbol 'wm_adsp2_ops' was not declared. Should it be static? sound/soc/codecs/wm_adsp.c:311:20: warning: symbol 'wm_halo_ops' was not declared. Should it be static? Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- v2: make variables static --- sound/soc/codecs/wm_adsp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 644aaf1..c41bffc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -306,9 +306,9 @@ #define HALO_MPU_VIO_ERR_SRC_MASK 0x00007fff #define HALO_MPU_VIO_ERR_SRC_SHIFT 0 -struct wm_adsp_ops wm_adsp1_ops; -struct wm_adsp_ops wm_adsp2_ops[]; -struct wm_adsp_ops wm_halo_ops; +static struct wm_adsp_ops wm_adsp1_ops; +static struct wm_adsp_ops wm_adsp2_ops[]; +static struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list; @@ -4407,13 +4407,13 @@ irqreturn_t wm_halo_wdt_expire(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_halo_wdt_expire); -struct wm_adsp_ops wm_adsp1_ops = { +static struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, .region_to_reg = wm_adsp_region_to_reg, }; -struct wm_adsp_ops wm_adsp2_ops[] = { +static struct wm_adsp_ops wm_adsp2_ops[] = { { .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, @@ -4474,7 +4474,7 @@ struct wm_adsp_ops wm_adsp2_ops[] = { }, }; -struct wm_adsp_ops wm_halo_ops = { +static struct wm_adsp_ops wm_halo_ops = { .sys_config_size = sizeof(struct wm_halo_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_halo_validate_version, -- 2.7.0

2 1

[alsa-devel] [PATCH 0/2] arm64: allwinner: a64: Rename hpvcc-supply to cpvdd-supply
by Chen-Yu Tsai 21 Mar '19

21 Mar '19

Hi everyone, This is a small series to fix the CPVDD regulator supply name for the A64 codec. The datasheet specifies the name "CPVDD" for the supply rail for the audio codec's internal headphone amp charge pump. Somehow I managed to write "CPVDD" in the commit message and the device tree binding, but used "HPVCC" (which is the name used by the A33) in the device tree files and the device driver. (facepalm) This feature was just introduced in v5.0-rc1, so if we could merge this as fixes for v5.0 that would be great. Thanks, and sorry for the blunder on my part. Regards ChenYu Chen-Yu Tsai (2): ASoC: sunxi: sun50i-codec-analog: Rename hpvcc regulator supply to cpvdd arm64: dts: allwinner: a64: Rename hpvcc-supply to cpvdd-supply arch/arm64/boot/dts/allwinner/sun50i-a64-bananapi-m64.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64-pine64.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64-pinebook.dts | 2 +- arch/arm64/boot/dts/allwinner/sun50i-a64-sopine.dtsi | 2 +- sound/soc/sunxi/sun50i-codec-analog.c | 4 ++-- 5 files changed, 6 insertions(+), 6 deletions(-) -- 2.20.1

3 5

[alsa-devel] [PATCH v2] ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286
by Jian-Hong Pan 21 Mar '19

21 Mar '19

Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot record sound from headset MIC. This patch adds the ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- v2: According to Realtek's suggestion, change the COEF 0x4f from 0xd429 to 0x5029. Thanks to Realtek! sound/pci/hda/patch_realtek.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 384719d5c44e..191830d4fa40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5687,6 +5687,7 @@ enum { ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE, ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, + ALC286_FIXUP_ACER_AIO_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6685,6 +6686,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC286_FIXUP_ACER_AIO_HEADSET_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x4f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5029 }, + { } + }, + .chained = true, + .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6701,9 +6712,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), - SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- 2.20.1

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[alsa-devel] [PATCH -next] ASoC: wm_adsp: Make some variables static
by Yue Haibing 21 Mar '19

21 Mar '19

From: YueHaibing <yuehaibing(a)huawei.com> Fix sparse warnings: sound/soc/codecs/wm_adsp.c:309:20: warning: symbol 'wm_adsp1_ops' was not declared. Should it be static? sound/soc/codecs/wm_adsp.c:310:20: warning: symbol 'wm_adsp2_ops' was not declared. Should it be static? sound/soc/codecs/wm_adsp.c:311:20: warning: symbol 'wm_halo_ops' was not declared. Should it be static? Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/wm_adsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 644aaf1..4ed587f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -306,9 +306,9 @@ #define HALO_MPU_VIO_ERR_SRC_MASK 0x00007fff #define HALO_MPU_VIO_ERR_SRC_SHIFT 0 -struct wm_adsp_ops wm_adsp1_ops; -struct wm_adsp_ops wm_adsp2_ops[]; -struct wm_adsp_ops wm_halo_ops; +static struct wm_adsp_ops wm_adsp1_ops; +static struct wm_adsp_ops wm_adsp2_ops[]; +static struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list; -- 2.7.0

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[alsa-devel] [PATCH RESEND] ARM: dts: exynos: Fix audio (microphone) routing on Odroid XU3
by Sylwester Nawrocki 20 Mar '19

20 Mar '19

The name of CODEC input widget to which microphone is connected through the "Headphone" jack is "IN12" not "IN1". This fixes microphone support on Odroid XU3. Cc: <stable(a)vger.kernel.org> # v4.14+ Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi b/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi index 3a37267e6134..c3c2d85267da 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi +++ b/arch/arm/boot/dts/exynos5422-odroidxu3-audio.dtsi @@ -22,7 +22,7 @@ "Headphone Jack", "HPL", "Headphone Jack", "HPR", "Headphone Jack", "MICBIAS", - "IN1", "Headphone Jack", + "IN12", "Headphone Jack", "Speakers", "SPKL", "Speakers", "SPKR", "I2S Playback", "Mixer DAI TX", -- 2.17.1

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[alsa-devel] [PATCH] ASoC: jz4740: Depend on MIPS instead of MACH_JZ4740
by Paul Cercueil 20 Mar '19

20 Mar '19

Depending on MACH_JZ4740 prevent us from creating a generic kernel that works on more than one MIPS board. Instead, we just depend on MIPS being set. Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> --- sound/soc/jz4740/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 1a354a6b6e87..b3f9c41b4319 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -1,6 +1,6 @@ config SND_JZ4740_SOC tristate "SoC Audio for Ingenic JZ4740 SoC" - depends on MACH_JZ4740 || COMPILE_TEST + depends on MIPS || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 2.11.0

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[alsa-devel] Applied "ASoC: tlv320aic32x4: Break out clock setting into separate function" to the asoc tree
by Mark Brown 20 Mar '19

20 Mar '19

The patch ASoC: tlv320aic32x4: Break out clock setting into separate function 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 bf31cbfbe25001036e1e096b1c260bf871766ea5 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Mon, 18 Mar 2019 20:37:44 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Break out clock setting into separate function Break the clock setting logic out from the main hw_params. It's rather large and unweildy and makes for a large function. This also better enables some of the following changes to the clock tree access in the driver. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 96f1526cb258..1aa8f5aa4225 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -696,17 +696,13 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -static int aic32x4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int aic32x4_setup_clocks(struct snd_soc_component *component, + unsigned int sample_rate, + unsigned int parent_rate) { - struct snd_soc_component *component = dai->component; - struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); - u8 iface1_reg = 0; - u8 dacsetup_reg = 0; int i; - i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + i = aic32x4_get_divs(parent_rate, sample_rate); if (i < 0) { printk(KERN_ERR "aic32x4: sampling rate not supported\n"); return i; @@ -763,6 +759,20 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AIC32X4_BCLKN, AIC32X4_BCLK_MASK, aic32x4_divs[i].blck_N); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + u8 iface1_reg = 0; + u8 dacsetup_reg = 0; + + aic32x4_setup_clocks(component, params_rate(params), aic32x4->sysclk); + switch (params_width(params)) { case 16: iface1_reg |= (AIC32X4_WORD_LEN_16BITS << -- 2.20.1

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Don't use register read to determine slave/master mode
by Sylwester Nawrocki 20 Mar '19

20 Mar '19

We can simplify the code by caching the CPU DAI master/slave information rather than reading previously set register bit. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ab471d550d17..9722940da6a4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -88,12 +88,6 @@ struct samsung_i2s_priv { struct platform_device *pdev; struct platform_device *pdev_sec; - /* Memory mapped SFR region */ - void __iomem *addr; - - /* Spinlock protecting access to the device's registers */ - spinlock_t lock; - /* Lock for cross interface checks */ spinlock_t pcm_lock; @@ -122,6 +116,15 @@ struct samsung_i2s_priv { /* The clock provider's data */ struct clk *clk_table[3]; struct clk_onecell_data clk_data; + + /* Spinlock protecting member fields below */ + spinlock_t lock; + + /* Memory mapped SFR region */ + void __iomem *addr; + + /* A flag indicating the I2S slave mode operation */ + bool slave_mode; }; /* Returns true if this is the 'overlay' stereo DAI */ @@ -130,15 +133,6 @@ static inline bool is_secondary(struct i2s_dai *i2s) return i2s->drv->id == SAMSUNG_I2S_ID_SECONDARY; } -/* If operating in SoC-Slave mode */ -static inline bool is_slave(struct i2s_dai *i2s) -{ - struct samsung_i2s_priv *priv = i2s->priv; - - u32 mod = readl(priv->addr + I2SMOD); - return (mod & (1 << priv->variant_regs->mss_off)) ? true : false; -} - /* If this interface of the controller is transmitting data */ static inline bool tx_active(struct i2s_dai *i2s) { @@ -715,6 +709,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, priv->addr + I2SMOD); + priv->slave_mode = (mod & mod_slave); spin_unlock_irqrestore(&priv->lock, flags); pm_runtime_put(dai->dev); @@ -917,7 +912,7 @@ static int config_setup(struct i2s_dai *i2s) set_rfs(i2s, rfs); /* Don't bother with PSR in Slave mode */ - if (is_slave(i2s)) + if (priv->slave_mode) return 0; if (!(priv->quirks & QUIRK_NO_MUXPSR)) { -- 2.17.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: Correct some minor issues
by Charles Keepax 20 Mar '19

20 Mar '19

The wm_adsp_ops structures should be static and correct two printf specifiers. Fixes: 170b1e123f38 ("ASoC: wm_adsp: Add support for new Halo core DSPs") Fixes: 4e08d50d1fb6 ("ASoC: wm_adsp: Factor out DSP specific operations") Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 644aaf1d27403..5e615fbddf9a7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -306,9 +306,9 @@ #define HALO_MPU_VIO_ERR_SRC_MASK 0x00007fff #define HALO_MPU_VIO_ERR_SRC_SHIFT 0 -struct wm_adsp_ops wm_adsp1_ops; -struct wm_adsp_ops wm_adsp2_ops[]; -struct wm_adsp_ops wm_halo_ops; +static struct wm_adsp_ops wm_adsp1_ops; +static struct wm_adsp_ops wm_adsp2_ops[]; +static struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list; @@ -2142,7 +2142,7 @@ static void wmfw_parse_id_header(struct wm_adsp *dsp, dsp->fw_id = be32_to_cpu(fw->id); dsp->fw_id_version = be32_to_cpu(fw->ver); - adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", + adsp_info(dsp, "Firmware: %x v%d.%d.%d, %d algorithms\n", dsp->fw_id, (dsp->fw_id_version & 0xff0000) >> 16, (dsp->fw_id_version & 0xff00) >> 8, dsp->fw_id_version & 0xff, nalgs); @@ -2155,7 +2155,7 @@ static void wmfw_v3_parse_id_header(struct wm_adsp *dsp, dsp->fw_id_version = be32_to_cpu(fw->ver); dsp->fw_vendor_id = be32_to_cpu(fw->vendor_id); - adsp_info(dsp, "Firmware: %x vendor: 0x%x v%d.%d.%d, %zu algorithms\n", + adsp_info(dsp, "Firmware: %x vendor: 0x%x v%d.%d.%d, %d algorithms\n", dsp->fw_id, dsp->fw_vendor_id, (dsp->fw_id_version & 0xff0000) >> 16, (dsp->fw_id_version & 0xff00) >> 8, dsp->fw_id_version & 0xff, @@ -4407,13 +4407,13 @@ irqreturn_t wm_halo_wdt_expire(int irq, void *data) } EXPORT_SYMBOL_GPL(wm_halo_wdt_expire); -struct wm_adsp_ops wm_adsp1_ops = { +static struct wm_adsp_ops wm_adsp1_ops = { .validate_version = wm_adsp_validate_version, .parse_sizes = wm_adsp1_parse_sizes, .region_to_reg = wm_adsp_region_to_reg, }; -struct wm_adsp_ops wm_adsp2_ops[] = { +static struct wm_adsp_ops wm_adsp2_ops[] = { { .sys_config_size = sizeof(struct wm_adsp_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, @@ -4474,7 +4474,7 @@ struct wm_adsp_ops wm_adsp2_ops[] = { }, }; -struct wm_adsp_ops wm_halo_ops = { +static struct wm_adsp_ops wm_halo_ops = { .sys_config_size = sizeof(struct wm_halo_system_config_xm_hdr), .parse_sizes = wm_adsp2_parse_sizes, .validate_version = wm_halo_validate_version, -- 2.11.0

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[alsa-devel] [PATCH 0/3] ASoC: stm32: dfsdm: fixes
by Olivier Moysan 20 Mar '19

20 Mar '19

This patch-set provides the following fixes for STM32 DFSDM: - manage multiple prepare - fix debugfs warnings on entry creation - add mmap support Olivier Moysan (3): ASoC: stm32: dfsdm: manage multiple prepare ASoC: stm32: dfsdm: fix debugfs warnings on entry creation ASoC: stm32: dfsdm: add mmap support sound/soc/stm/stm32_adfsdm.c | 40 +++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) -- 2.7.4

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[alsa-devel] [RFC PATCH asoc] ASoC: wm_adsp: wm_halo_ops can be static
by kbuild test robot 20 Mar '19

20 Mar '19

Fixes: 170b1e123f38 ("ASoC: wm_adsp: Add support for new Halo core DSPs") Signed-off-by: kbuild test robot <lkp(a)intel.com> --- wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 644aaf1..33b39c6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -308,7 +308,7 @@ struct wm_adsp_ops wm_adsp1_ops; struct wm_adsp_ops wm_adsp2_ops[]; -struct wm_adsp_ops wm_halo_ops; +static struct wm_adsp_ops wm_halo_ops; struct wm_adsp_buf { struct list_head list;

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[alsa-devel] [PATCH 1/2] ASoC: lochnagar: Add device tree binding document
by Charles Keepax 20 Mar '19

20 Mar '19

Lochnagar is an evaluation and development board for Cirrus Logic Smart CODEC and Amp devices. It allows the connection of most Cirrus Logic devices on mini-cards, as well as allowing connection of various application processor systems to provide a full evaluation platform. This driver supports the board controller chip on the Lochnagar board. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- .../devicetree/bindings/sound/cirrus,lochnagar.txt | 39 ++++++++++++++++++++++ 1 file changed, 39 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt new file mode 100644 index 0000000000000..fba5fb53a2f90 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,lochnagar.txt @@ -0,0 +1,39 @@ +Cirrus Logic Lochnagar Audio Development Board + +Lochnagar is an evaluation and development board for Cirrus Logic +Smart CODEC and Amp devices. It allows the connection of most Cirrus +Logic devices on mini-cards, as well as allowing connection of +various application processor systems to provide a full evaluation +platform. Audio system topology, clocking and power can all be +controlled through the Lochnagar, allowing the device under test +to be used in a variety of possible use cases. + +This binding document describes the binding for the audio portion +of the driver. + +This binding must be part of the Lochnagar MFD binding: + [4] ../mfd/cirrus,lochnagar.txt + +Required properties: + + - compatible : One of the following strings: + "cirrus,lochnagar2-soundcard" + + - #sound-dai-cells : Must be set to 1. + + - clocks : Contains an entry for each entry in clock-names. + - clock-names : Must include the following clocks: + "ln-sound-mclk" Master clock source for the sound card, should + normally be set to LOCHNAGAR_SOUNDCARD_MCLK provided by the + Lochnagar clock driver. + +Example: + +lochnagar-sc { + compatible = "cirrus,lochnagar2-soundcard"; + + #sound-dai-cells = <1>; + + clocks = <&lochnagar_clk LOCHNAGAR_SOUNDCARD_MCLK>; + clock-names = "ln-soundcard-mclk"; +}; -- 2.11.0

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