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[alsa-devel] [PATCH] sound: soc-pcm: add a check to avoid NULL pointer dereference
by Kangjie Lu 15 Mar '19

15 Mar '19

In case debugfs_create_u32 fails, the fix frees memory and returns an error to notify callers. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> --- sound/soc/soc-pcm.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0d5ec68a1e50..a7d88ed8ebb5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1231,15 +1231,21 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", - stream ? "capture" : "playback", fe->dai_link->name, - stream ? "<-" : "->", be->dai_link->name); - #ifdef CONFIG_DEBUG_FS - if (fe->debugfs_dpcm_root) + if (fe->debugfs_dpcm_root) { dpcm->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644, fe->debugfs_dpcm_root, &dpcm->state); + if (!dpcm->debugfs_state) { + kfree(dpcm); + return -ENOMEM; + } + } #endif + + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", + stream ? "capture" : "playback", fe->dai_link->name, + stream ? "<-" : "->", be->dai_link->name); + return 1; } -- 2.17.1

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[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 14 Mar '19

14 Mar '19

The patch ASoC: rsnd: src: fix compiler 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 825b22e5961a47ba7387d3ee6729956d6bb3b495 Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 11:34:53 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index db81e066b92e..79716725f67b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -135,7 +135,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -144,7 +144,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -153,7 +153,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -162,7 +162,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -171,7 +171,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -180,7 +180,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix clang warning without CONFIG_PM
by Arnd Bergmann 14 Mar '19

14 Mar '19

Building with clang shows a variable that is only used by the suspend/resume functions but defined outside of their #ifdef block: sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted We commonly fix these by marking the PM functions as __maybe_unused, but here that would grow the davinci_mcasp structure, so instead add another #ifdef here. Fixes: 1cc0c054f380 ("ASoC: davinci-mcasp: Convert the context save/restore to use array") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/ti/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a3a67a8f0f54..9fbc759fdefe 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -45,6 +45,7 @@ #define MCASP_MAX_AFIFO_DEPTH 64 +#ifdef CONFIG_PM static u32 context_regs[] = { DAVINCI_MCASP_TXFMCTL_REG, DAVINCI_MCASP_RXFMCTL_REG, @@ -68,6 +69,7 @@ struct davinci_mcasp_context { u32 *xrsr_regs; /* for serializer configuration */ bool pm_state; }; +#endif struct davinci_mcasp_ruledata { struct davinci_mcasp *mcasp; -- 2.20.0

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[alsa-devel] [PATCH 1/3] ASoC: ti: fix davinci_mcasp_probe dependencies
by Arnd Bergmann 14 Mar '19

14 Mar '19

The SND_SOC_DAVINCI_MCASP driver can use either edma or sdma as a back-end, and it takes the presence of the respective dma engine drivers in the configuration as an indication to which ones should be built. However, this is flawed in multiple ways: - With CONFIG_TI_EDMA=m and CONFIG_SND_SOC_DAVINCI_MCASP=y, is enabled as =m, and we get a link error: sound/soc/ti/davinci-mcasp.o: In function `davinci_mcasp_probe': davinci-mcasp.c:(.text+0x930): undefined reference to `edma_pcm_platform_register' - When CONFIG_SND_SOC_DAVINCI_MCASP=m has already been selected by another driver, the same link error appears even if CONFIG_TI_EDMA is disabled There are possibly other issues here, but it seems that the only reasonable solution is to always build both SND_SOC_TI_EDMA_PCM and SND_SOC_TI_SDMA_PCM as a dependency here. Both are fairly small and do not have any other compile-time dependencies, so the cost is very small, and makes the configuration stage much more consistent. Fixes: f2055e145f29 ("ASoC: ti: Merge davinci and omap directories") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/ti/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 4bf3c15d4e51..ee7c202c69b7 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -21,8 +21,8 @@ config SND_SOC_DAVINCI_ASP config SND_SOC_DAVINCI_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - select SND_SOC_TI_EDMA_PCM if TI_EDMA - select SND_SOC_TI_SDMA_PCM if DMA_OMAP + select SND_SOC_TI_EDMA_PCM + select SND_SOC_TI_SDMA_PCM help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: -- 2.20.0

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[alsa-devel] [PATCH 0/3] ASoC: stm32: i2s: add minor improvements
by Olivier Moysan 14 Mar '19

14 Mar '19

Add following minor changes in STM32 I2S driver: - Change trigger traces - Improve channel capabilities handling - Use default dai name Olivier Moysan (3): ASoC: stm32: i2s: change trigger traces ASoC: stm32: i2s: improve channel capabilities handling ASoC: stm32: i2s: use default dai name sound/soc/stm/stm32_i2s.c | 25 +++++++++---------------- 1 file changed, 9 insertions(+), 16 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: ams-delta: remove duplicate 'const'
by Arnd Bergmann 14 Mar '19

14 Mar '19

clang points out that SOC_ENUM_SINGLE_EXT_DECL() contains a 'const' modifier already, so adding another one does not make it more const: sound/soc/ti/ams-delta.c:203:14: error: duplicate 'const' declaration specifier [-Werror,-Wduplicate-decl-specifier] static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, ^ include/sound/soc.h:351:2: note: expanded from macro 'SOC_ENUM_SINGLE_EXT_DECL' const struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/ti/ams-delta.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 4dce494dfbd3..b9611db14c86 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -200,7 +200,7 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, return 0; } -static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, +static SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, ams_delta_audio_mode); static const struct snd_kcontrol_new ams_delta_audio_controls[] = { -- 2.20.0

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[alsa-devel] [PATCH 1/2] ASoC: axi-i2s: let both capture and playback be optional
by Luca Ceresoli 14 Mar '19

14 Mar '19

Both the capture and playback channels are optional in the axi_i2s IP block. Reflect this in the driver by enabling only the channel(s) that have a DMA. Signed-off-by: Luca Ceresoli <luca(a)lucaceresoli.net> --- sound/soc/adi/axi-i2s.c | 68 +++++++++++++++++++++++++++++------------ 1 file changed, 48 insertions(+), 20 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 4c23381727a1..273c543e8ff3 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -43,6 +43,9 @@ struct axi_i2s { struct clk *clk; struct clk *clk_ref; + bool has_capture; + bool has_playback; + struct snd_soc_dai_driver dai_driver; struct snd_dmaengine_dai_dma_data capture_dma_data; @@ -136,8 +139,10 @@ static int axi_i2s_dai_probe(struct snd_soc_dai *dai) { struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); - snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, - &i2s->capture_dma_data); + snd_soc_dai_init_dma_data( + dai, + i2s->has_playback ? &i2s->playback_dma_data : NULL, + i2s->has_capture ? &i2s->capture_dma_data : NULL); return 0; } @@ -151,18 +156,6 @@ static const struct snd_soc_dai_ops axi_i2s_dai_ops = { static struct snd_soc_dai_driver axi_i2s_dai = { .probe = axi_i2s_dai_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE, - }, .ops = &axi_i2s_dai_ops, .symmetric_rates = 1, }; @@ -178,6 +171,19 @@ static const struct regmap_config axi_i2s_regmap_config = { .max_register = AXI_I2S_REG_STATUS, }; +static void axi_i2s_parse_of(struct axi_i2s *i2s, const struct device_node *np) +{ + struct property *dma_names; + const char *dma_name; + + of_property_for_each_string(np, "dma-names", dma_names, dma_name) { + if (strcmp(dma_name, "rx") == 0) + i2s->has_capture = true; + if (strcmp(dma_name, "tx") == 0) + i2s->has_playback = true; + } +} + static int axi_i2s_probe(struct platform_device *pdev) { struct resource *res; @@ -191,6 +197,8 @@ static int axi_i2s_probe(struct platform_device *pdev) platform_set_drvdata(pdev, i2s); + axi_i2s_parse_of(i2s, pdev->dev.of_node); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(base)) @@ -213,13 +221,29 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) return ret; - i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO; - i2s->playback_dma_data.addr_width = 4; - i2s->playback_dma_data.maxburst = 1; + if (i2s->has_playback) { + axi_i2s_dai.playback.channels_min = 2; + axi_i2s_dai.playback.channels_max = 2; + axi_i2s_dai.playback.rates = SNDRV_PCM_RATE_KNOT; + axi_i2s_dai.playback.formats = + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE; + + i2s->playback_dma_data.addr = res->start + AXI_I2S_REG_TX_FIFO; + i2s->playback_dma_data.addr_width = 4; + i2s->playback_dma_data.maxburst = 1; + } + + if (i2s->has_capture) { + axi_i2s_dai.capture.channels_min = 2; + axi_i2s_dai.capture.channels_max = 2; + axi_i2s_dai.capture.rates = SNDRV_PCM_RATE_KNOT; + axi_i2s_dai.capture.formats = + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE; - i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO; - i2s->capture_dma_data.addr_width = 4; - i2s->capture_dma_data.maxburst = 1; + i2s->capture_dma_data.addr = res->start + AXI_I2S_REG_RX_FIFO; + i2s->capture_dma_data.addr_width = 4; + i2s->capture_dma_data.maxburst = 1; + } i2s->ratnum.num = clk_get_rate(i2s->clk_ref) / 2 / AXI_I2S_BITS_PER_FRAME; i2s->ratnum.den_step = 1; @@ -240,6 +264,10 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + dev_info(&pdev->dev, "probed, capture %s, playback %s\n", + i2s->has_capture ? "enabled" : "disabled", + i2s->has_playback ? "enabled" : "disabled"); + return 0; err_clk_disable: -- 2.17.1

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[alsa-devel] Applied "ASoC: nau8810: use 64-bit arithmetic instead of 32-bit" to the asoc tree
by Mark Brown 14 Mar '19

14 Mar '19

The patch ASoC: nau8810: use 64-bit arithmetic instead of 32-bit 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 b517229ca2f7836125be58997808f2803f9ebc86 Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Wed, 13 Mar 2019 14:47:18 +0800 Subject: [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit 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> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- 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; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: nau8810: fix the typo of function name
by John Hsu 14 Mar '19

14 Mar '19

Correct the typo at the function name. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index e7fd0b2..39512a7 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -493,7 +493,7 @@ static int nau8810_set_sysclk(struct snd_soc_dai *dai, return 0; } -static int nau88l0_calc_pll(unsigned int pll_in, +static int nau8810_calc_pll(unsigned int pll_in, unsigned int fs, struct nau8810_pll *pll_param) { u64 f2, f2_max, pll_ratio; @@ -542,7 +542,7 @@ static int nau8810_set_pll(struct snd_soc_dai *codec_dai, int pll_id, int ret, fs; fs = freq_out / 256; - ret = nau88l0_calc_pll(freq_in, fs, pll_param); + ret = nau8810_calc_pll(freq_in, fs, pll_param); if (ret < 0) { dev_err(nau8810->dev, "Unsupported input clock %d\n", freq_in); return ret; -- 2.18.0

2 2

[alsa-devel] [PATCH] ASoC: nau8810: automatic selecting BCLK in I2S master mode
by John Hsu 14 Mar '19

14 Mar '19

The driver will select correct BCLK automatically according to BCLK and FS information in I2S master mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 832657720a5f..125e205e6687 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -667,6 +667,24 @@ static int nau8810_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct nau8810 *nau8810 = snd_soc_component_get_drvdata(component); int val_len = 0, val_rate = 0, ret = 0; + unsigned int ctrl_val, bclk_fs, bclk_div; + + /* Select BCLK configuration if the codec as master. */ + regmap_read(nau8810->regmap, NAU8810_REG_CLOCK, &ctrl_val); + if (ctrl_val & NAU8810_CLKIO_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / params_rate(params); + if (bclk_fs <= 32) + bclk_div = NAU8810_BCLKDIV_8; + else if (bclk_fs <= 64) + bclk_div = NAU8810_BCLKDIV_4; + else if (bclk_fs <= 128) + bclk_div = NAU8810_BCLKDIV_2; + else + return -EINVAL; + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_BCLKSEL_MASK, bclk_div); + } switch (params_width(params)) { case 16: -- 2.18.0

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[alsa-devel] Support two Dell Wyse platform
by Kailang 14 Mar '19

14 Mar '19

Hi Takashi, Attach patch was for Wyse NB and Wyse AIO platform. BR, Kailang

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[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headset MIC of Acer AIO with ALC286
by Jian-Hong Pan 14 Mar '19

14 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> --- 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 a63670013943..b3fbad5a680f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5695,6 +5695,7 @@ enum { ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, ALC255_FIXUP_ACER_HEADSET_MIC, ALC295_FIXUP_CHROME_BOOK, + ALC286_FIXUP_ACER_AIO_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6677,6 +6678,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC225_FIXUP_HEADSET_JACK }, + [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, 0xd429 }, + { } + }, + .chained = true, + .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6693,9 +6704,9 @@ 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

2 1

Re: [alsa-devel] opening rawmidi on user ports
by Clemens Ladisch 14 Mar '19

14 Mar '19

AlGrenadine wrote: > Le 13/03/2019 à 14:37, Clemens Ladisch a écrit : >> Load the snd-virmidi module to get a bridge between raw MIDI ports and sequencer ports. > > Ok then what would be the snd_rawmidi_open argument for name of virmidi port ? See "amidi -l". Regards, Clemens

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek: merge alc_fixup_headset_jack to alc295_fixup_chromebook
by Jaroslav Kysela 14 Mar '19

14 Mar '19

The ALC225_FIXUP_HEADSET_JACK fixup can be merged to alc295_fixup_chromebook. There are no other users for ALC225_FIXUP_HEADSET_JACK other than the chromebook hardware. Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 45 +++++++++++------------------------ 1 file changed, 14 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a63670013943..eb4018469250 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5491,7 +5491,7 @@ static void alc_headset_btn_callback(struct hda_codec *codec, jack->jack->button_state = report; } -static void alc_fixup_headset_jack(struct hda_codec *codec, +static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5501,6 +5501,16 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, alc_headset_btn_callback); snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, SND_JACK_HEADSET, alc_headset_btn_keymap); + 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; case HDA_FIXUP_ACT_INIT: switch (codec->core.vendor_id) { @@ -5521,26 +5531,6 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, } } -static void alc295_fixup_chromebook(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - 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; - } -} - static void alc_fixup_disable_mic_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5690,11 +5680,10 @@ enum { ALC294_FIXUP_ASUS_MIC, ALC294_FIXUP_ASUS_HEADSET_MIC, ALC294_FIXUP_ASUS_SPK, - ALC225_FIXUP_HEADSET_JACK, + ALC295_FIXUP_CHROME_BOOK, ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, ALC255_FIXUP_ACER_HEADSET_MIC, - ALC295_FIXUP_CHROME_BOOK, }; static const struct hda_fixup alc269_fixups[] = { @@ -6637,9 +6626,9 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, - [ALC225_FIXUP_HEADSET_JACK] = { + [ALC295_FIXUP_CHROME_BOOK] = { .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_headset_jack, + .v.func = alc295_fixup_chromebook, }, [ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -6671,12 +6660,6 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC }, - [ALC295_FIXUP_CHROME_BOOK] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc295_fixup_chromebook, - .chained = true, - .chain_id = ALC225_FIXUP_HEADSET_JACK - }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { -- 2.20.1

3 2

[alsa-devel] [PATCH] ALSA: hda/realtek: merge alc295_fixup_chromebook to alc_fixup_headset_jack
by Jaroslav Kysela 14 Mar '19

14 Mar '19

The ALC295_FIXUP_CHROME_BOOK fixup can be merged to alc_fixup_headset_jack unless some other model will reference ALC225_FIXUP_HEADSET_JACK in an incompatible way. Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 39 ++++++++++------------------------- 1 file changed, 11 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a63670013943..41ac066c6ed7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5501,6 +5501,16 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, alc_headset_btn_callback); snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, SND_JACK_HEADSET, alc_headset_btn_keymap); + 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; case HDA_FIXUP_ACT_INIT: switch (codec->core.vendor_id) { @@ -5521,26 +5531,6 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, } } -static void alc295_fixup_chromebook(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - 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; - } -} - static void alc_fixup_disable_mic_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5694,7 +5684,6 @@ enum { ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, ALC255_FIXUP_ACER_HEADSET_MIC, - ALC295_FIXUP_CHROME_BOOK, }; static const struct hda_fixup alc269_fixups[] = { @@ -6671,12 +6660,6 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC }, - [ALC295_FIXUP_CHROME_BOOK] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc295_fixup_chromebook, - .chained = true, - .chain_id = ALC225_FIXUP_HEADSET_JACK - }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7089,7 +7072,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DUMMY_LINEOUT_VERB, .name = "alc255-dummy-lineout"}, {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, - {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-sense-combo"}, + {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-sense-combo"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.20.1

2 1

[alsa-devel] ChromeBook machine will lose MIC JD
by Kailang 14 Mar '19

14 Mar '19

Hi Takashi, Attach patch will solve the issue for Mic JD lose. BR, Kailang

3 8

[alsa-devel] [PATCH] ALSA: hda: hdmi - add Icelake support
by Jaroslav Kysela 14 Mar '19

14 Mar '19

This is just a port of the ASoC Icelake HDMI codec code to the legacy HDA driver with some cleanups. ASoC commit 019033c854a20e10f691f6cc0e897df8817d9521: "ASoC: Intel: hdac_hdmi: add Icelake support" Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Bard liao <bard.liao(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 67 +++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 73d7042ff884..8b3ac690efa3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -57,10 +57,11 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ ((codec)->core.vendor_id == 0x80862800)) #define is_cannonlake(codec) ((codec)->core.vendor_id == 0x8086280c) +#define is_icelake(codec) ((codec)->core.vendor_id == 0x8086280f) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) || is_geminilake(codec) \ - || is_cannonlake(codec) + || is_kabylake(codec) || is_geminilake(codec) \ + || is_cannonlake(codec) || is_icelake(codec)) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) #define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) @@ -181,6 +182,8 @@ struct hdmi_spec { struct hdac_chmap chmap; hda_nid_t vendor_nid; + const int *port_map; + int port_num; }; #ifdef CONFIG_SND_HDA_COMPONENT @@ -2418,12 +2421,11 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -#define INTEL_VENDOR_NID 0x08 -#define INTEL_GLK_VENDOR_NID 0x0B -#define INTEL_GET_VENDOR_VERB 0xf81 -#define INTEL_SET_VENDOR_VERB 0x781 -#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ -#define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ +#define INTEL_GET_VENDOR_VERB 0xf81 +#define INTEL_GET_VENDOR_VERB 0xf81 +#define INTEL_SET_VENDOR_VERB 0x781 +#define INTEL_EN_DP12 0x02 /* enable DP 1.2 features */ +#define INTEL_EN_ALL_PIN_CVTS 0x01 /* enable 2nd & 3rd pins and convertors */ static void intel_haswell_enable_all_pins(struct hda_codec *codec, bool update_tree) @@ -2503,11 +2505,29 @@ static int intel_base_nid(struct hda_codec *codec) static int intel_pin2port(void *audio_ptr, int pin_nid) { - int base_nid = intel_base_nid(audio_ptr); + struct hda_codec *codec = audio_ptr; + struct hdmi_spec *spec = codec->spec; + int base_nid, i; - if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3)) - return -1; - return pin_nid - base_nid + 1; /* intel port is 1-based */ + if (!spec->port_num) { + base_nid = intel_base_nid(codec); + if (WARN_ON(pin_nid < base_nid || pin_nid >= base_nid + 3)) + return -1; + return pin_nid - base_nid + 1; /* intel port is 1-based */ + } + + /* + * looking for the pin number in the mapping table and return + * the index which indicate the port number + */ + for (i = 0; i < spec->port_num; i++) { + if (pin_nid == spec->port_map[i]) + return i + 1; + } + + /* return -1 if pin number exceeds our expectation */ + codec_info(codec, "Can't find the HDMI/DP port for pin %d\n", pin_nid); + return -1; } static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) @@ -2608,7 +2628,8 @@ static int parse_intel_hdmi(struct hda_codec *codec) } /* Intel Haswell and onwards; audio component with eld notifier */ -static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid) +static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, + const int *port_map, int port_num) { struct hdmi_spec *spec; int err; @@ -2620,6 +2641,8 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid) codec->dp_mst = true; spec->dyn_pcm_assign = true; spec->vendor_nid = vendor_nid; + spec->port_map = port_map; + spec->port_num = port_num; intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); @@ -2638,12 +2661,23 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid) static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, INTEL_VENDOR_NID); + return intel_hsw_common_init(codec, 0x08, NULL, 0); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, INTEL_GLK_VENDOR_NID); + return intel_hsw_common_init(codec, 0x0b, NULL, 0); +} + +static int patch_i915_icl_hdmi(struct hda_codec *codec) +{ + /* + * pin to port mapping table where the value indicate the pin number and + * the index indicate the port number with 1 base. + */ + static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb}; + + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); } /* Intel Baytrail and Braswell; with eld notifier */ @@ -3886,6 +3920,7 @@ HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), @@ -3899,7 +3934,7 @@ HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), -HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), +HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- 2.20.1

3 4

[alsa-devel] [asoc:for-5.2 14/16] ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
by kbuild test robot 14 Mar '19

14 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 20b83421e8eea43e28cb7aeea3e3f865f86aa69d commit: b517229ca2f7836125be58997808f2803f9ebc86 [14/16] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit 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 b517229ca2f7836125be58997808f2803f9ebc86 # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All errors (new ones prefixed by >>): >> ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined! --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-5.2 14/16] ERROR: "__udivdi3" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
by kbuild test robot 14 Mar '19

14 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 20b83421e8eea43e28cb7aeea3e3f865f86aa69d commit: b517229ca2f7836125be58997808f2803f9ebc86 [14/16] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit config: i386-randconfig-m1-03132108 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout b517229ca2f7836125be58997808f2803f9ebc86 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> ERROR: "__udivdi3" [sound/soc/codecs/snd-soc-nau8810.ko] undefined! --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: pcm: Fix function name in kernel-doc comment
by Ricardo Biehl Pasquali 13 Mar '19

13 Mar '19

Signed-off-by: Ricardo Biehl Pasquali <pasqualirb(a)gmail.com> --- include/sound/pcm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 465d7d033..18bd8c3ea 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -750,7 +750,7 @@ static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *r } /** - * snd_pcm_playback_avail - Get the available (readable) space for capture + * snd_pcm_capture_avail - Get the available (readable) space for capture * @runtime: PCM runtime instance * * Result is between 0 ... (boundary - 1) -- 2.19.1

2 1

Re: [alsa-devel] opening rawmidi on user ports
by Clemens Ladisch 13 Mar '19

13 Mar '19

AlGrenadine wrote: > it seems ALSA seq looses sysex messages. Is there a buffer size which can be set ? Messages from where to where? Can you check with aseqdump or "amidi --dump"? Messages that are too large for the internal buffer are split into multiple parts. Regards, Clemens

1 0

[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 13 Mar '19

13 Mar '19

The patch ASoC: rsnd: src: fix compiler 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 825b22e5961a47ba7387d3ee6729956d6bb3b495 Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 11:34:53 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index db81e066b92e..79716725f67b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -135,7 +135,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -144,7 +144,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -153,7 +153,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -162,7 +162,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -171,7 +171,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -180,7 +180,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: nau8810: fix the issue of widget with prefixed name
by John Hsu 13 Mar '19

13 Mar '19

The driver changes the stream name of DAC and ADC to avoid the issue of widget with prefixed name. When the machine adds prefixed name for codec, the stream name of DAI may not find the widgets. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 39512a78a556..832657720a5f 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -411,9 +411,9 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], ARRAY_SIZE(nau8810_mono_mixer_controls)), - SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + SND_SOC_DAPM_DAC("DAC", "Playback", NAU8810_REG_POWER3, NAU8810_DAC_EN_SFT, 0), - SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + SND_SOC_DAPM_ADC("ADC", "Capture", NAU8810_REG_POWER2, NAU8810_ADC_EN_SFT, 0), SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, NAU8810_NSPK_EN_SFT, 0, NULL, 0), -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: samsung: odroid: Fix clock configuration for 44100 sample rate
by Sylwester Nawrocki 13 Mar '19

13 Mar '19

After commit fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") the audio root clock frequency is configured improperly for 44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead of 22579000 Hz. This results in a too low value of the PSR clock divider in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g. 1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2). Fix this by increasing the correction passed to clk_set_rate() to take into account inaccuracy of the EPLL frequency properly. Fixes: fbeec965b8d1c ("ASoC: samsung: odroid: Fix 32000 sample rate handling") Reported-by: JaeChul Lee <jcsing.lee(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 694512f980fd..1dc54c4206f0 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -91,11 +91,11 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, return ret; /* - * We add 1 to the rclk_freq value in order to avoid too low clock + * We add 2 to the rclk_freq value in order to avoid too low clock * frequency values due to the EPLL output frequency not being exact * multiple of the audio sampling rate. */ - rclk_freq = params_rate(params) * rfs + 1; + rclk_freq = params_rate(params) * rfs + 2; ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: nau8810: use 64-bit arithmetic instead of 32-bit" to the asoc tree
by Mark Brown 13 Mar '19

13 Mar '19

The patch ASoC: nau8810: use 64-bit arithmetic instead of 32-bit 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 b517229ca2f7836125be58997808f2803f9ebc86 Mon Sep 17 00:00:00 2001 From: John Hsu <KCHSU0(a)nuvoton.com> Date: Wed, 13 Mar 2019 14:47:18 +0800 Subject: [PATCH] ASoC: nau8810: use 64-bit arithmetic instead of 32-bit 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> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- 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; -- 2.20.1

1 0

Re: [alsa-devel] [PATCH 2/7] drm/edid: Allow to ignore the audio EDID data
by Ville Syrjälä 13 Mar '19

13 Mar '19

On Tue, Mar 05, 2019 at 10:12:40AM +0100, Maxime Ripard wrote: > On Mon, Mar 04, 2019 at 03:05:31PM -0500, Alex Deucher wrote: > > On Mon, Mar 4, 2019 at 2:53 PM Eric Anholt <eric(a)anholt.net> wrote: > > > > > > Maxime Ripard <maxime.ripard(a)bootlin.com> writes: > > > > > > > In some cases, in order to accomodate with displays with poor EDIDs, we > > > > need to ignore that the monitor alledgedly supports audio output and > > > > disable the audio output. > > > > > > > > Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> > > > > --- > > > > drivers/gpu/drm/drm_edid.c | 8 ++++++++ > > > > 1 file changed, 8 insertions(+) > > > > > > > > diff --git a/drivers/gpu/drm/drm_edid.c b/drivers/gpu/drm/drm_edid.c > > > > index 990b1909f9d7..c0258b011bb2 100644 > > > > --- a/drivers/gpu/drm/drm_edid.c > > > > +++ b/drivers/gpu/drm/drm_edid.c > > > > @@ -4190,6 +4190,11 @@ bool drm_detect_hdmi_monitor(struct edid *edid) > > > > } > > > > EXPORT_SYMBOL(drm_detect_hdmi_monitor); > > > > > > > > +static bool ignore_edid_audio = false; > > > > +module_param(ignore_edid_audio, bool, 0644); > > > > +MODULE_PARM_DESC(ignore_edid_audio, > > > > + "Ignore the EDID and always consider that a monitor doesn't have audio capabilities"); > > > > + > > > > /** > > > > * drm_detect_monitor_audio - check monitor audio capability > > > > * @edid: EDID block to scan > > > > @@ -4209,6 +4214,9 @@ bool drm_detect_monitor_audio(struct edid *edid) > > > > bool has_audio = false; > > > > int start_offset, end_offset; > > > > > > > > + if (ignore_edid_audio) > > > > + goto end; > > > > + > > > > edid_ext = drm_find_cea_extension(edid); > > > > if (!edid_ext) > > > > goto end; > > > > > > It looks like the motivation for the original flag on Raspberry Pi was > > > "I've got a non-audio monitor, but the system comes up trying to play > > > audio to HDMI instead of the analog jack". Do we have some way for DRM > > > to communicate to ALSA that this is not the right place to try to play > > > audio by default? > > > > Apparently not. We have users using debug knobs in our drivers to > > disable display audio because ALSA defaults to that rather than other > > audio. > > I guess one way to do this would be to register the card only when an > audio-capable monitor is connected instead of doing this at probe > time. I'm not sure how convenient it is for userspace though. We already provide the ELD to alsa. I'm pretty sure pulseaudio uses that stuff somehow to figure out whether to play audio over HDMI. But since I don't use pulseaudio myself I can't be 100% sure. Cc:ing Takashi/alsa folks for confirmation. -- Ville Syrjälä Intel

4 5

[alsa-devel] [PATCH] ALSA: hda - add more quirks for HP Z2 G4 and HP Z240
by Jaroslav Kysela 13 Mar '19

13 Mar '19

From: Jaroslav Kysela <jkysela(a)redhat.com> Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and HP Z240 models. Reported-by: Jeff Burrell <jeff.burrell(a)hp.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_conexant.c | 3 +++ sound/pci/hda/patch_realtek.c | 6 ++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a4ee7656d9ee..fb65ad31e86c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -936,6 +936,9 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8458, "HP Z2 G4 mini premium", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c8413d44973c..476c3ee31dc8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6764,11 +6764,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), - SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.20.1

2 3

[alsa-devel] opening rawmidi on user ports
by AlGrenadine 13 Mar '19

13 Mar '19

Hi, I use btmidi-server from the bluez stack, which creates user ports to send and receive midi via bluetooth. I can't manager which parameter string i must provide to snd_rawmidi_open function, can you help ? (of course the user ports don't appear in amidi -l) I only have a client and port ids (128:0) when using aconnect -l Thanks

3 2

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable headset MIC of Acer TravelMate X514-51T with ALC255
by Jian-Hong Pan 13 Mar '19

13 Mar '19

The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the internal DMIC uses another module - snd_soc_skl as the driver. We still need the NID 0x1a in the quirk to enable the headset MIC. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c8413d44973c..8debd661e6f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5673,6 +5673,7 @@ enum { ALC225_FIXUP_HEADSET_JACK, ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, + ALC255_FIXUP_ACER_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6639,6 +6640,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_LENOVO_HEADPHONE_NOISE }, + [ALC255_FIXUP_ACER_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11130 }, + { 0x1a, 0x90a60140 }, /* use as internal mic */ + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6658,6 +6669,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 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), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- 2.20.1

2 1

Re: [alsa-devel] [PATCH] Fix speakers on Acer Predator Helios 500 Ryzen laptops
by Takashi Iwai 13 Mar '19

13 Mar '19

On Tue, 05 Mar 2019 00:38:19 +0100, Bernhard Rosenkraenzer wrote: > > On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers > don't work out of the box. > > The problem can be worked around with hdajackretask, remapping the > "Black Headphone, Right side" pin (0x21) to the Internal speaker. > > This patch adds a quirk to change this mapping by default. > > Signed-off-by: Bernhard Rosenkraenzer <bero(a)lindev.ch> Applied, thanks. Takashi

1 1

[alsa-devel] [PATCH] ALSA: hda/tegra: avoid build error without CONFIG_PM
by Arnd Bergmann 13 Mar '19

13 Mar '19

The #ifdef protection around the PM functions is wrong, leading to a failed reference in some configurations: sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend': sound/pci/hda/hda_tegra.c:273:2: error: implicit declaration of function 'hda_tegra_disable_clocks'; did you mean 'hda_tegra_enable_clocks'? [-Werror=implicit-function-declaration] Better remove the #ifdefs entirely and rely on the compiler silently dropping unused functions marked __maybe_unused. Fixes: 707e0759f2f4 ("ALSA: hda/tegra: implement runtime suspend/resume") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/hda_tegra.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index dbd8da5685cb..3d68f9ef7694 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -230,7 +229,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) /* * power management */ -static int hda_tegra_suspend(struct device *dev) +static int __maybe_unused hda_tegra_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); int rc; @@ -243,7 +242,7 @@ static int hda_tegra_suspend(struct device *dev) return 0; } -static int hda_tegra_resume(struct device *dev) +static int __maybe_unused hda_tegra_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); int rc; @@ -255,10 +254,8 @@ static int hda_tegra_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM -static int hda_tegra_runtime_suspend(struct device *dev) +static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; @@ -275,7 +272,7 @@ static int hda_tegra_runtime_suspend(struct device *dev) return 0; } -static int hda_tegra_runtime_resume(struct device *dev) +static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; @@ -292,7 +289,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) -- 2.20.0

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Re: [alsa-devel] [PATCH] ALSA: usx2y: Fix potential NULL pointer dereference
by Takashi Iwai 13 Mar '19

13 Mar '19

On Tue, 05 Mar 2019 00:00:02 +0100, Aditya Pakki wrote: > > usb_alloc_urb can fail due to kmalloc failure and push the error > upstream. Further this can cause a NULL pointer dereference in > init_pipe_urbs. This patch avoids such a scenario. > > Signed-off-by: Aditya Pakki <pakki001(a)umn.edu> Applied, thanks. Takashi

1 0

[alsa-devel] [PATCH 0/1] sound/hda/hdac_stream: Avoid NULL pointer dereference
by Mariusz Ceier 13 Mar '19

13 Mar '19

Seems like commit 9b6f7e7a296e17990aae298c809b001e99ddd151 introduced NULL pointer dereference for ca0132 codec. When ca0132 loads firmware, snd_hdac_stream_start is called with azx_dev->substream being NULL. This patch calls snd_hdac_get_stream_stripe_ctl only when azx_dev->substream is not NULL. Even if I'm not sure if this is correct, since it might be that ca0132 codec does something wrong, with this change NULL pointer dereference doesn't happen and ca0132 works again on my system with Recon3Di. Mariusz Ceier (1): sound/hda/hdac_stream: Avoid NULL pointer dereference sound/hda/hdac_stream.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) -- 2.21.0

2 2

[alsa-devel] [RFC PATCH 1/4] alsa: make hw_params negotiation infrastructure 'bclk_ratio aware'
by Sven Van Asbroeck 13 Mar '19

13 Mar '19

Negotiation seems to work ok, but bclk_ratio is exposed to userspace via snd_pcm_hw_params, which is not acceptable. Constrain bclk_ratio by: - cpu dai capabilities && rules - codec dai capabilities && rules - minimum bclk_ratio is sample_width * channels In hw_params_choose(), pick the smallest supported bclk_ratio, which should correspond to the most efficient solution. If cpu and codec dais do not specify or constrain supported bclk_ratios, alsa will pick sample_width * channels. Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- include/sound/pcm.h | 11 +++++++++++ include/sound/soc.h | 2 ++ include/uapi/sound/asound.h | 5 +++-- sound/core/pcm_native.c | 34 +++++++++++++++++++++++++++++++++- sound/soc/soc-pcm.c | 8 ++++++++ 5 files changed, 57 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index d6bd3caf6878..71ac7e8de23d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -56,6 +56,8 @@ struct snd_pcm_hardware { unsigned int periods_min; /* min # of periods */ unsigned int periods_max; /* max # of periods */ size_t fifo_size; /* fifo size in bytes */ + unsigned int bclk_ratio_min; /* min bclk ratio for wire format */ + unsigned int bclk_ratio_max; /* max bclk ratio for wire format */ }; struct snd_pcm_substream; @@ -980,6 +982,15 @@ static inline unsigned int params_buffer_bytes(const struct snd_pcm_hw_params *p return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min; } +/** + * params_bclk_ratio - Get the bclk_ratio from the hw params + * @p: hw params + */ +static inline unsigned int params_bclk_ratio(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BCLK_RATIO)->min; +} + int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); int snd_interval_list(struct snd_interval *i, unsigned int count, const unsigned int *list, unsigned int mask); diff --git a/include/sound/soc.h b/include/sound/soc.h index e665f111b0d2..96d669423688 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -732,6 +732,8 @@ struct snd_soc_pcm_stream { unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ unsigned int sig_bits; /* number of bits of content */ + unsigned int bclk_ratio_min; /* min bclk ratio for wire format */ + unsigned int bclk_ratio_max; /* max bclk ratio for wire format */ }; /* SoC audio ops */ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 404d4b9ffe76..c3ea94eaaa77 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -371,8 +371,9 @@ typedef int snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_BUFFER_SIZE 17 /* Size of buffer in frames */ #define SNDRV_PCM_HW_PARAM_BUFFER_BYTES 18 /* Size of buffer in bytes */ #define SNDRV_PCM_HW_PARAM_TICK_TIME 19 /* Approx tick duration in us */ +#define SNDRV_PCM_HW_PARAM_BCLK_RATIO 20 /* bclk_ratio for wire format */ #define SNDRV_PCM_HW_PARAM_FIRST_INTERVAL SNDRV_PCM_HW_PARAM_SAMPLE_BITS -#define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME +#define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_BCLK_RATIO #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ #define SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER (1<<1) /* export buffer */ @@ -399,7 +400,7 @@ struct snd_pcm_hw_params { struct snd_mask mres[5]; /* reserved masks */ struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; - struct snd_interval ires[9]; /* reserved intervals */ + struct snd_interval ires[8]; /* reserved intervals */ unsigned int rmask; /* W: requested masks */ unsigned int cmask; /* R: changed masks */ unsigned int info; /* R: Info flags for returned setup */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 818dff1de545..23dbe43a6691 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -516,6 +516,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int err; + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_BCLK_RATIO); params->info = 0; params->fifo_size = 0; @@ -525,6 +526,12 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, params->rate_num = 0; params->rate_den = 0; } + /* + * if left zero (not empty), assume userspace is oblivious, and + * completely flexible + */ + if (snd_interval_single(r) && snd_interval_min(r) == 0) + snd_interval_any(r); err = constrain_mask_params(substream, params); if (err < 0) @@ -610,7 +617,8 @@ static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, - * min rate, min period time, max buffer size, min tick time + * min rate, min period time, max buffer size, min tick time, + * min bclk_ratio * * Return: Zero if successful, or a negative error code on failure. */ @@ -626,6 +634,7 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_TICK_TIME, + SNDRV_PCM_HW_PARAM_BCLK_RATIO, -1 }; const int *v; @@ -2100,6 +2109,18 @@ static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, return snd_mask_refine(mask, &m); } +static int snd_pcm_hw_rule_bclk_ratio(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval i; + struct snd_interval *ratios = hw_param_interval(params, SNDRV_PCM_HW_PARAM_BCLK_RATIO); + snd_interval_any(&i); + i.openmax = 1; + i.min = params_channels(params) * params_width(params); + i.integer = 1; + return snd_interval_refine(ratios, &i); +} + static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -2180,12 +2201,18 @@ int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)); snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)); snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_FRAME_BITS)); + snd_interval_setinteger(constrs_interval(constrs, SNDRV_PCM_HW_PARAM_BCLK_RATIO)); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, snd_pcm_hw_rule_format, NULL, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BCLK_RATIO, + snd_pcm_hw_rule_bclk_ratio, NULL, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_sample_bits, NULL, SNDRV_PCM_HW_PARAM_FORMAT, @@ -2341,6 +2368,11 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BCLK_RATIO, + hw->bclk_ratio_min, hw->bclk_ratio_max); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_buffer_bytes_max, substream, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 03f36e534050..747026595634 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -381,6 +381,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) struct snd_soc_pcm_stream *cpu_stream; unsigned int chan_min = 0, chan_max = UINT_MAX; unsigned int rate_min = 0, rate_max = UINT_MAX; + unsigned int bclk_ratio_min = 0, bclk_ratio_max = UINT_MAX; unsigned int rates = UINT_MAX; u64 formats = ULLONG_MAX; int i; @@ -413,6 +414,8 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) codec_stream = &codec_dai_drv->capture; chan_min = max(chan_min, codec_stream->channels_min); chan_max = min(chan_max, codec_stream->channels_max); + bclk_ratio_min = max(bclk_ratio_min, codec_stream->bclk_ratio_min); + bclk_ratio_max = min_not_zero(bclk_ratio_max, codec_stream->bclk_ratio_max); rate_min = max(rate_min, codec_stream->rate_min); rate_max = min_not_zero(rate_max, codec_stream->rate_max); formats &= codec_stream->formats; @@ -443,6 +446,11 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_min = max(hw->rate_min, rate_min); hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); hw->rate_max = min_not_zero(hw->rate_max, rate_max); + + hw->bclk_ratio_min = max(hw->bclk_ratio_min, cpu_stream->bclk_ratio_min); + hw->bclk_ratio_min = max(hw->bclk_ratio_min, bclk_ratio_min); + hw->bclk_ratio_max = min_not_zero(hw->bclk_ratio_max, cpu_stream->bclk_ratio_max); + hw->bclk_ratio_max = min_not_zero(hw->bclk_ratio_max, bclk_ratio_max); } static int soc_pcm_components_close(struct snd_pcm_substream *substream, -- 2.17.1

6 22

[alsa-devel] [PATCH v2 0/5] use extended audio dmac register
by Jiada Wang 13 Mar '19

13 Mar '19

According to user reference manual for R-CAR H3, M3-W and M3-N SoCs, in order to access busif4 ~ busif7, extended audio dmac registers (PDMASAREn, PDMADAREn, PDMACHCREn) need to be used, rather than basic audio dmac registers (PDMASARn, PDMADARn, PDMACHCRn) This patch set updates H3 (= r8a7795), M3-W (= r8a7796) and M3-N (=r8a77965) to use extended audio dmac registers R-CAR E3 only has extended register set and is already using it, rename register set name to indicate this. --- v2: update rsnd driver to use extended register set when available rename R-CAR E3 audio dmac regisetr v1: initial version Jiada Wang (5): arm64: dts: renesas: r8a7795: add extended audio dmac register arm64: dts: renesas: r8a7796: add extended audio dmac register arm64: dts: renesas: r8a77965: add extended audio dmac register arm64: dts: renesas: r8a77990: rename audio dmac register ASoC: rsnd: dma: use extended audio dmac registers when available arch/arm64/boot/dts/renesas/r8a7795.dtsi | 5 +++-- arch/arm64/boot/dts/renesas/r8a7796.dtsi | 5 +++-- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 5 +++-- arch/arm64/boot/dts/renesas/r8a77990.dtsi | 4 ++-- sound/soc/sh/rcar/dma.c | 5 ++++- 5 files changed, 15 insertions(+), 9 deletions(-) -- 2.19.2

1 4

[alsa-devel] [PATCH 1/2] ASoC: audio-graph-card: test memory allocation
by Pierre-Louis Bossart 12 Mar '19

12 Mar '19

devm_kcalloc() return value needs to be checked. Fixes: 1e4771a62fd7a ('ASoC: audio-graph-card: add link_info') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/generic/audio-graph-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index bb12351330e8..ba65b48c10a7 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -670,7 +670,7 @@ static int graph_probe(struct platform_device *pdev) 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) + if (!dai_props || !dai_link || !dais || !cconf) return -ENOMEM; /* -- 2.17.1

2 5

[alsa-devel] [PATCH] media: Drop superfluous PCM preallocation error checks
by Takashi Iwai 12 Mar '19

12 Mar '19

snd_pcm_lib_preallocate_pages() and co always succeed, so the error check is simply redundant. Drop it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is a piece I overlooked in the previous patchset: https://www.spinics.net/lists/alsa-devel/msg87223.html Media people, please give ACK if this is OK. Then I'll merge it together with other relevant patches. Thanks! drivers/media/pci/solo6x10/solo6x10-g723.c | 4 +--- drivers/media/pci/tw686x/tw686x-audio.c | 3 ++- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/drivers/media/pci/solo6x10/solo6x10-g723.c b/drivers/media/pci/solo6x10/solo6x10-g723.c index 2cc05a9d57ac..a16242a9206f 100644 --- a/drivers/media/pci/solo6x10/solo6x10-g723.c +++ b/drivers/media/pci/solo6x10/solo6x10-g723.c @@ -360,13 +360,11 @@ static int solo_snd_pcm_init(struct solo_dev *solo_dev) ss; ss = ss->next, i++) sprintf(ss->name, "Camera #%d Audio", i); - ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), G723_PERIOD_BYTES * PERIODS, G723_PERIOD_BYTES * PERIODS); - if (ret < 0) - return ret; solo_dev->snd_pcm = pcm; diff --git a/drivers/media/pci/tw686x/tw686x-audio.c b/drivers/media/pci/tw686x/tw686x-audio.c index a28329698e20..fb0e7573b5ae 100644 --- a/drivers/media/pci/tw686x/tw686x-audio.c +++ b/drivers/media/pci/tw686x/tw686x-audio.c @@ -301,11 +301,12 @@ static int tw686x_snd_pcm_init(struct tw686x_dev *dev) ss; ss = ss->next, i++) snprintf(ss->name, sizeof(ss->name), "vch%u audio", i); - return snd_pcm_lib_preallocate_pages_for_all(pcm, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(dev->pci_dev), TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX, TW686X_AUDIO_PAGE_MAX * AUDIO_DMA_SIZE_MAX); + return 0; } static void tw686x_audio_dma_free(struct tw686x_dev *dev, -- 2.16.4

3 2

[alsa-devel] [PATCH] ASoC: samsung: i2s: Fix DAPM routes for capture stream
by Sylwester Nawrocki 11 Mar '19

11 Mar '19

This patch sets missing stream_name of capture part of the DAI driver so we can define DAPM routing properly also for the capture stream. While at it "Playback" suffix is added to the playback stream names to clearly identify playback/capture. Together with related dts patch this fixes NULL pointer dereference when opening ALSA device for recording on Odroid XU3. Fixes: 64aba9bca5bd ("ASoC: samsung: i2s: Add widgets and routes for DPCM support") Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 4231001226f4..ab471d550d17 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1130,11 +1130,11 @@ static const struct snd_soc_dapm_widget samsung_i2s_widgets[] = { }; static const struct snd_soc_dapm_route samsung_i2s_dapm_routes[] = { - { "Playback Mixer", NULL, "Primary" }, - { "Playback Mixer", NULL, "Secondary" }, + { "Playback Mixer", NULL, "Primary Playback" }, + { "Playback Mixer", NULL, "Secondary Playback" }, { "Mixer DAI TX", NULL, "Playback Mixer" }, - { "Playback Mixer", NULL, "Mixer DAI RX" }, + { "Primary Capture", NULL, "Mixer DAI RX" }, }; static const struct snd_soc_component_driver samsung_i2s_component = { @@ -1155,7 +1155,8 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, int num_dais) { static const char *dai_names[] = { "samsung-i2s", "samsung-i2s-sec" }; - static const char *stream_names[] = { "Primary", "Secondary" }; + static const char *stream_names[] = { "Primary Playback", + "Secondary Playback" }; struct snd_soc_dai_driver *dai_drv; struct i2s_dai *dai; int i; @@ -1201,6 +1202,7 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, dai_drv->capture.channels_max = 2; dai_drv->capture.rates = i2s_dai_data->pcm_rates; dai_drv->capture.formats = SAMSUNG_I2S_FMTS; + dai_drv->capture.stream_name = "Primary Capture"; return 0; } -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 11 Mar '19

11 Mar '19

The patch ASoC: rsnd: src: fix compiler 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 825b22e5961a47ba7387d3ee6729956d6bb3b495 Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 11:34:53 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index db81e066b92e..79716725f67b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -135,7 +135,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -144,7 +144,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -153,7 +153,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -162,7 +162,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -171,7 +171,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -180,7 +180,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

1 0

[alsa-devel] [PATCH 0/3] ASoC: hda: capture-related fixes
by Rander Wang 11 Mar '19

11 Mar '19

There are several sources of errors fixed in this series, such as mismatch between TX and RX mask used to store the stream_tag, the use of hw_param fixups not properly handled in the hdac_hda codec and registers set on capture when they should only be used on playback. The first patch fixes a issue caused by tx|rx_mask for TDM. The Second one fixes the mismatch between BE dai format and hda codec format. The last one removes the register of stream id setting on capture. this series was tested successfully with the Intel Skylake driver on a Skylake Dell XPS13 (same as Linus), KabyLake NUC, Whiskylake laptop and Apollolake LeafHill board. The two soc-pcm and hdac_hda fixes are needed by the SOF driver as well, but SOF-specific patches will be provided in the next batch and are not included here Rander Wang (3): ASoC:soc-pcm:fix a codec fixup issue in TDM case ASoC:hdac_hda:use correct format to setup hda codec ASoC:intel:skl:fix a simultaneous playback & capture issue on hda platform sound/soc/codecs/hdac_hda.c | 53 ++++++++++++++++++++++++++++----------- sound/soc/codecs/hdac_hda.h | 1 + sound/soc/intel/skylake/skl-pcm.c | 19 ++++++++++---- sound/soc/soc-pcm.c | 7 ++++-- 4 files changed, 59 insertions(+), 21 deletions(-) -- 2.14.1

3 6

[alsa-devel] Applied "ASoC:hdac_hda:use correct format to setup hda codec" to the asoc tree
by Mark Brown 11 Mar '19

11 Mar '19

The patch ASoC:hdac_hda:use correct format to setup hda codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 03d0aa4d4fddce4a5d865d819a4d98bfc3d451e6 Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang(a)linux.intel.com> Date: Fri, 8 Mar 2019 16:38:58 +0800 Subject: [PATCH] ASoC:hdac_hda:use correct format to setup hda codec The current implementation of the hdac_hda codec results in zero-valued samples on capture and noise with headset playback when SOF is used on platforms with an on-board HDaudio codec. This is root-caused to SOF using be_hw_params_fixup, and the prepare() call using invalid runtime fields to determine the format. This patch moves the format handling to the hw_params() callback, as done already for hdac_hdmi, to make sure the fixed-up information is taken into account but keeps the codec initialization in prepare() as the stream_tag is only available at that time. Moving everything in the prepare() callback is possible but the code is less elegant so this two-step solution was chosen. The solution was tested with the SST driver with no regressions, and all the issues with SOF playback and capture are solved. Signed-off-by: Rander Wang <rander.wang(a)linux.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/codecs/hdac_hda.c | 53 +++++++++++++++++++++++++++---------- sound/soc/codecs/hdac_hda.h | 1 + 2 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index ffecdaaa8cf2..f889d94c8e3c 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -38,6 +38,9 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -50,6 +53,7 @@ static const struct snd_soc_dai_ops hdac_hda_dai_ops = { .startup = hdac_hda_dai_open, .shutdown = hdac_hda_dai_close, .prepare = hdac_hda_dai_prepare, + .hw_params = hdac_hda_dai_hw_params, .hw_free = hdac_hda_dai_hw_free, .set_tdm_slot = hdac_hda_dai_set_tdm_slot, }; @@ -139,6 +143,39 @@ static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int hdac_hda_dai_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 hdac_hda_priv *hda_pvt; + unsigned int format_val; + unsigned int maxbps; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + maxbps = dai->driver->playback.sig_bits; + else + maxbps = dai->driver->capture.sig_bits; + + hda_pvt = snd_soc_component_get_drvdata(component); + format_val = snd_hdac_calc_stream_format(params_rate(params), + params_channels(params), + params_format(params), + maxbps, + 0); + if (!format_val) { + dev_err(dai->dev, + "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + params_rate(params), params_channels(params), + params_format(params), maxbps); + + return -EINVAL; + } + + hda_pvt->pcm[dai->id].format_val[substream->stream] = format_val; + return 0; +} + static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,10 +199,9 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct snd_pcm_runtime *runtime = substream->runtime; struct hdac_device *hdev; - struct hda_pcm_stream *hda_stream; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; @@ -179,19 +215,8 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hda_stream->maxbps, - 0); - if (!format_val) { - dev_err(&hdev->dev, - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - return -EINVAL; - } - stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, stream, format_val, substream); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index e444ef593360..6b1bd4f428e7 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -8,6 +8,7 @@ struct hdac_hda_pcm { int stream_tag[2]; + unsigned int format_val[2]; }; struct hdac_hda_priv { -- 2.20.1

1 0

[alsa-devel] [PATCH v3] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 11 Mar '19

11 Mar '19

The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Use irq spin lock version, since the lock may be used in interrupts. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 40 +++++++++++++++++++++++++++++++++------- 3 files changed, 36 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db60..1e2be35 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764d..d887576 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e8..30df83a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1213,6 +1213,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { struct snd_soc_dpcm *dpcm; + unsigned long flags; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1228,8 +1229,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1275,6 +1278,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; + unsigned long flags; for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", @@ -1294,8 +1298,10 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); } } @@ -1547,10 +1553,13 @@ int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + unsigned long flags; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2571,6 +2580,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret; + unsigned long flags; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); @@ -2640,11 +2650,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -3220,7 +3232,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + unsigned long flags; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3244,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3265,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + unsigned long flags; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3278,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0; + break; + } } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3301,6 +3325,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; struct snd_soc_dpcm *dpcm; ssize_t offset = 0; + unsigned long flags; /* FE state */ offset += snprintf(buf + offset, size - offset, @@ -3328,6 +3353,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3374,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); out: return offset; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: nau8824: fix the issue of the widget with prefix name
by John Hsu 11 Mar '19

11 Mar '19

The driver has two issues when machine add prefix name for codec. (1)The stream name of DAI can't find the AIF widgets. (2)The drivr can enable/disalbe the MICBIAS and SAR widgets. The patch will fix these issues caused by prefixed name added. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 46 +++++++++++++++++++++++++++++++------- 1 file changed, 38 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 87ed3dc496dc..5ab05e75edea 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -681,8 +681,8 @@ static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCR", NULL, NAU8824_REG_ANALOG_ADC_2, NAU8824_ADCR_EN_SFT, 0), - SND_SOC_DAPM_AIF_OUT("AIFTX", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACL", NULL, NAU8824_REG_RDAC, NAU8824_DACL_EN_SFT, 0), @@ -831,6 +831,36 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } +static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + const char *prefix = dapm->component->name_prefix; + char prefixed_pin[80]; + + if (prefix) { + snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", + prefix, pin); + snd_soc_dapm_disable_pin(dapm, prefixed_pin); + } else { + snd_soc_dapm_disable_pin(dapm, pin); + } +} + +static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + const char *prefix = dapm->component->name_prefix; + char prefixed_pin[80]; + + if (prefix) { + snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", + prefix, pin); + snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); + } else { + snd_soc_dapm_force_enable_pin(dapm, pin); + } +} + static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -839,8 +869,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -870,8 +900,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_force_enable_pin(dapm, "SAR"); + nau8824_dapm_enable_pin(nau8824, "MICBIAS"); + nau8824_dapm_enable_pin(nau8824, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -882,8 +912,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: stm32: i2s: fix registers declaration in regmap
by Olivier Moysan 11 Mar '19

11 Mar '19

- Declare SR as volatile, as it is changed by hardware. - Remove TXDR from readable and volatile register list, as it is intended for write accesses only. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index c18e068c1a0d..952634258b02 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -278,7 +278,6 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) case STM32_I2S_CFG2_REG: case STM32_I2S_IER_REG: case STM32_I2S_SR_REG: - case STM32_I2S_TXDR_REG: case STM32_I2S_RXDR_REG: case STM32_I2S_CGFR_REG: return true; @@ -290,7 +289,7 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) static bool stm32_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case STM32_I2S_TXDR_REG: + case STM32_I2S_SR_REG: case STM32_I2S_RXDR_REG: return true; default: -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 11 Mar '19

11 Mar '19

The patch ASoC: rsnd: src: fix compiler 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 399706df420ea8bc8b31283a919e6475f737d0ea Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 15:15:54 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 45096a66c074..585ffba0244b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -134,7 +134,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -143,7 +143,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -152,7 +152,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -161,7 +161,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -170,7 +170,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -179,7 +179,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

1 0

[alsa-devel] [PATCH v11 0/5] Media Device Allocator API
by Shuah Khan 11 Mar '19

11 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 rested 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. - I am sharing the test plan used for testing resource sharing which could serve as a regression test plan. 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. - 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. 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 (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 | 4 + drivers/media/media-dev-allocator.c | 142 +++++++++++ drivers/media/usb/au0828/au0828-core.c | 190 ++++++++++---- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 53 ++++ 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, 865 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

2 6

[alsa-devel] [PATCH v2] ASoC: fsl: sai: Fix underrrun for playback stream start
by Daniel Baluta 11 Mar '19

11 Mar '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> For [dir=playback, cmd = PCM_TRIGGER_START], fsl_sai_trigger does the following: * Enable DMA request * Enable Transmitter Now, because DMA doesn't copy data to TDR fast enough we hit TX FIFO underrun error: [ 37.175974] fsl-sai 308b0000.sai: isr: Transmit underrun detected The explanation for this can be found in SAI documentation for example chapter 16.16.3.5.2 FIFO pointers of the i.MX8DXP reference manual [1] " If the Transmit FIFO is empty, then to avoid a FIFO underrun, the Transmit Data Register must be written at least 3 bit clocks before the start of the next unmasked word. Before enabling the transmitter, the Transmit FIFO should be initialized with data (since after the transmitter is enabled, the transmitter will start a new frame, and if no data is in the FIFO, then the transmitter will immediately give an error). " [1] https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: - as per Fabio's request clarify the issue and the solution - move words writing before enabling DMA request sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2d376e3a3c80..a739baa99c1a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -522,7 +522,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int channels = substream->runtime->channels; u32 xcsr, count = 100; + int i; /* * Asynchronous mode: Clear SYNC for both Tx and Rx. @@ -542,6 +544,11 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + for (i = 0; tx && i < channels; i++) + regmap_write(sai->regmap, FSL_SAI_TDR, 0x0); + if (tx) + udelay(10); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: fsl: sai: Reduce underrun when stream starts
by Daniel Baluta 08 Mar '19

08 Mar '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> Write initial words into SAI FIFO to reduce the underrun error. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9e0872f73d..99edbba4e8fa 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -503,7 +503,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int channels = substream->runtime->channels; u32 xcsr, count = 100; + int i; /* * Asynchronous mode: Clear SYNC for both Tx and Rx. @@ -526,6 +528,11 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); + for (i = 0; tx && i < channels; i++) + regmap_write(sai->regmap, FSL_SAI_TDR, 0x0); + if (tx) + udelay(10); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); regmap_update_bits(sai->regmap, FSL_SAI_TCSR, -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by mac.chiang＠intel.com 08 Mar '19

08 Mar '19

From: Mac Chiang <mac.chiang(a)intel.com> s0ix entry failure is root cause by default speaker/vi sense path activate all the time that result in the audio IPs are not power gated(checked by pch_ip_power_gating_status). Adding the run time speaker control ensure playback widgets are going to free and off eventually. With this func, counter works in s0ix subsequently (checked by slp_s0_residency_usc). 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> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 50 ++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 723a493..e01ca9a 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -195,8 +195,58 @@ 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, MAXIM_DEV0_NAME) && + strcmp(name, MAXIM_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, -- 2.7.4

3 8

[alsa-devel] [PATCH Resend] ASoC: rt5682: Check JD status when system resume
by shumingf＠realtek.com 08 Mar '19

08 Mar '19

From: Shuming Fan <shumingf(a)realtek.com> The IRQ function may not work when system suspend. We remove snd_soc_dapm_force_enable_pin function call to make sure the bias off when idle and run into suspend/resume function. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9d5acd2d04ab..b7e2e9937e6d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -910,13 +910,20 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); unsigned int val, count; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, RT5682_PWR_VREF2); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, RT5682_PWR_CBJ); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); @@ -944,8 +951,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_dapm_disable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, 0); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, 0); rt5682->jack_type = 0; } @@ -1591,8 +1600,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1627,9 +1634,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, - RT5682_PWR_CBJ_BIT, 0, NULL, 0), - /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, ARRAY_SIZE(rt5682_rec1_l_mix)), @@ -1792,17 +1796,13 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*Vref*/ {"MICBIAS1", NULL, "Vref1"}, - {"MICBIAS1", NULL, "Vref2"}, {"MICBIAS2", NULL, "Vref1"}, - {"MICBIAS2", NULL, "Vref2"}, {"CLKDET SYS", NULL, "CLKDET"}, {"IN1P", NULL, "LDO2"}, {"BST1 CBJ", NULL, "IN1P"}, - {"BST1 CBJ", NULL, "CBJ Power"}, - {"CBJ Power", NULL, "Vref2"}, {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, {"RECMIX1L", NULL, "RECMIX1L Power"}, @@ -1912,9 +1912,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Capless"}, {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, - {"HP Amp", NULL, "CBJ Power"}, {"HP Amp", NULL, "Vref1"}, - {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HPOL Playback"}, @@ -2363,6 +2361,8 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); + rt5682_irq(0, rt5682); + return 0; } #else -- 2.21.0

1 0

[alsa-devel] Question about devm_get_clk_from_child()
by Kuninori Morimoto 08 Mar '19

08 Mar '19

Hi Stephen I noticed that this commit breaks my ALSA Sound playback 4472287a3b2f52f4aa53f294ccb74392dde4e07d ("clk: Introduce of_clk_get_hw_from_clkspec()") I debuged this issue, and noticed that devm_get_clk_from_child() can't get clk my pseudo DT code is like this deviceA: devA { ... device = <&devB> }; deviceB: devB { clocks = <xxxx> }; I could get clock on deviceB driver // dev = deviceB clk = clk_get(dev, NULL); clk is "clocks = <xxx>" But, I couldn't get clock from deviceA driver // node = device = devB node = ... // dev = deviceA clk = devm_get_clk_from_child(dev, node, NULL); My understanding is these can get same clock. But, am I wrong ?? Best regards --- Kuninori Morimoto

2 5

[alsa-devel] [PATCH] ASoC: soc-pcm: only apply symmetry if more than one stream is active
by Russell King 07 Mar '19

07 Mar '19

ALSA documentation indicates that the hw_params() (and prepare()) callbacks can be made multiple times during the life of an individual stream. It is not specified whether hw_free() will be called, nor is it specified whether the parameters will be the same as the last call. Since symmetry is supposed to be applied between different streams, apply symmetry only when we have more than one activation of the CPU DAI. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- Untested, depends on "ASoC: soc-pcm: make symmetry work again" sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2a4246dbe55f..3c7c3f79742e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1010,9 +1010,11 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; + if (cpu_dai->active > 1) { + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto component_err; + } /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); -- 2.7.4

1 0

[alsa-devel] [PATCH v2] ASoC: soc-pcm: make symmetry work again
by Russell King 07 Mar '19

07 Mar '19

Commit 957ce0c6b8a1 ("ASoC: soc-pcm: check symmetry after hw_params") moved the check for symmetry after we have set the cpu_dai's rate, channels and sample width to those in the params. The symmetry check verifies that the cpu dai's stored parameters are the same as those in the params. Since we have only just stored these parameters, this makes the symmetry checks always pass. Fix this by moving the symmetry check before we store the parameters, but after the parameters have been fixed up. Fixes: 957ce0c6b8a1 ("ASoC: soc-pcm: check symmetry after hw_params") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- Not tested: this is something I stumbled over while reading the code. v2: also fixes the incorrect whitespace indentation copied from the original patch. sound/soc/soc-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 08ee163504e8..2a4246dbe55f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1010,15 +1010,16 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto component_err; + /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); cpu_dai->channels = params_channels(params); cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; out: mutex_unlock(&rtd->pcm_mutex); return ret; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc-pcm: make symmetry work again
by Russell King 07 Mar '19

07 Mar '19

Commit 957ce0c6b8a1 ("ASoC: soc-pcm: check symmetry after hw_params") moved the check for symmetry after we have set the cpu_dai's rate, channels and sample width to those in the params. The symmetry check verifies that the cpu dai's stored parameters are the same as those in the params. Since we have only just stored these parameters, this makes the symmetry checks always pass. Fix this by moving the symmetry check before we store the parameters, but after the parameters have been fixed up. Fixes: 957ce0c6b8a1 ("ASoC: soc-pcm: check symmetry after hw_params") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- Not tested: this is something I stumbled over while reading the code. sound/soc/soc-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 08ee163504e8..dc3c3e31d2ef 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1010,15 +1010,16 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } component = NULL; + ret = soc_pcm_params_symmetry(substream, params); + if (ret) + goto component_err; + /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); cpu_dai->channels = params_channels(params); cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); - ret = soc_pcm_params_symmetry(substream, params); - if (ret) - goto component_err; out: mutex_unlock(&rtd->pcm_mutex); return ret; -- 2.7.4

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Boards: Add Maxim98373 support
by Jenny TC 07 Mar '19

07 Mar '19

Reuse kbl_da7219_max98927 machine driver to support max98373 v1: First patchset V2: Updated commit message and Acked-by Jenny TC (1): ASoC: Intel: Boards: Add Maxim98373 support sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 203 +++++++++++++++++++--- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 19 ++ 3 files changed, 200 insertions(+), 23 deletions(-) -- 1.9.1

4 4

[alsa-devel] ChromeBook machine will lose MIC JD(again)
by Kailang 07 Mar '19

07 Mar '19

Hi Takashi, Attach patch will solve the issue for Mic JD lose. BR, Kailang

1 0

[alsa-devel] [PATCH linux-next v1 0/2] misc fix in src.c
by Jiada Wang 07 Mar '19

07 Mar '19

two issues were found due to linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") this patch-set address these two issues Jiada Wang (2): ASoC: rsnd: src: Avoid a potential deadlock ASoC: rsnd: src: fix compiler warnings sound/soc/sh/rcar/core.c | 3 +++ sound/soc/sh/rcar/rsnd.h | 5 +++++ sound/soc/sh/rcar/src.c | 21 +++++++-------------- 3 files changed, 15 insertions(+), 14 deletions(-) -- 2.19.2

2 4

[alsa-devel] [PATCH v2] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 07 Mar '19

07 Mar '19

The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 33 ++++++++++++++++++++++++++------- 3 files changed, 29 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db60..1e2be35 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764d..d887576 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e8..9020a06 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1228,8 +1228,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + spin_lock(&fe->card->dpcm_lock); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + spin_unlock(&fe->card->dpcm_lock); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1294,8 +1296,10 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); + spin_unlock(&fe->card->dpcm_lock); kfree(dpcm); } } @@ -1548,9 +1552,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2646,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3220,7 +3228,9 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3239,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3260,9 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3272,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3328,6 +3346,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3367,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 07 Mar '19

07 Mar '19

the dpcm get from fe_clients/be_clients may be free before use possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 33 ++++++++++++++++++++++++++------- 3 files changed, 29 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db60..1e2be35 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764d..d887576 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e8..9020a06 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1228,8 +1228,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + spin_lock(&fe->card->dpcm_lock); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + spin_unlock(&fe->card->dpcm_lock); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1294,8 +1296,10 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); + spin_unlock(&fe->card->dpcm_lock); kfree(dpcm); } } @@ -1548,9 +1552,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2646,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3220,7 +3228,9 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3239,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3260,9 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3272,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3328,6 +3346,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3367,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 07 Mar '19

07 Mar '19

the dpcm get from fe_clients/be_clients may be free before use possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 33 ++++++++++++++++++++++++++------- 3 files changed, 29 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db60..1e2be35 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764d..d887576 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e8..e99346e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1228,8 +1228,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + spin_lock(&fe->card->dpcm_lock); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + spin_unlock(&fe->card->dpcm_lock); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1294,8 +1296,10 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); + spin_unlock(&fe->card->dpcm_lock); kfree(dpcm); } } @@ -1548,9 +1552,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2646,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3220,7 +3228,9 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3239,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3260,9 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3272,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0 + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3328,6 +3346,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3367,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 1.9.1

1 0

[alsa-devel] [RFC PATCH v2] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 06 Mar '19

06 Mar '19

the dpcm get from fe_clients/be_clients may be free before use possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch off mixer between FE2 -> BE Thread 2: pcm_stop FE1 Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 31 ++++++++++++++++++++++++------- 3 files changed, 27 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db605955b..1e2be35ed36f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764dba147..d88757659729 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e82dea4..f31eaf58e750 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,9 +1294,11 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); + spin_unlock(&fe->card->dpcm_lock); } } @@ -1548,9 +1550,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2644,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3220,7 +3226,9 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3237,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3258,9 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3270,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0 + break; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3328,6 +3344,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3365,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 2.18.0

3 3

[alsa-devel] [PATCH] ALSA: hda: set the 2nd headphone's name to be Headphone2
by Hui Wang 06 Mar '19

06 Mar '19

Recently I saw two Headphones on some HP machines, the 1st Headphone comes from the Front Headphone Jack, the 2nd one comes from the Front Headset Jack, and the current HDA driver will set the same name to them and with different index, just like below: 'Headphone Playback Volume' 'Headphone Playback Switch' 'Headphone Playback Volume' index 1 'Headphone Playback Switch' index 1 'Front Headphone Jack' 'Front Headphone Jack' index 1 This can't be handled by pulseaudio, and pulseaduio already has a path analog-output-headphones-2.conf to handle the 2nd headphone, now we set the 2nd headphone's name to "Headphone2", leave the other headphones unchanged. The change of hda_auto_parser.c will change the name of Jack. The change of hda_generic.c will change the name of Volume and Switch. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- The alsa-info.txt before applying this patch: https://pastebin.ubuntu.com/p/CrPrHRykgT/ The alsa-info.txt after applying this patch: https://pastebin.ubuntu.com/p/9VbkyvMBkW/ sound/pci/hda/hda_auto_parser.c | 11 +++++++++-- sound/pci/hda/hda_generic.c | 2 ++ 2 files changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index df0d636145f8..748c62a9c5eb 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -714,6 +714,8 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); const char *name = NULL; int i; + int ret; + int index; bool hdmi; if (indexp) @@ -729,8 +731,13 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, return fill_audio_out_name(codec, nid, cfg, "Speaker", label, maxlen, indexp); case AC_JACK_HP_OUT: - return fill_audio_out_name(codec, nid, cfg, "Headphone", - label, maxlen, indexp); + ret = fill_audio_out_name(codec, nid, cfg, "Headphone", + label, maxlen, &index); + if (index == 1) + snprintf(label, maxlen, "%s", "Headphone2"); + if (indexp) + *indexp = index; + return ret; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: hdmi = is_hdmi_cfg(def_conf); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 4095cd7c56c6..2a34e8a6e2e1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2175,6 +2175,8 @@ static int create_extra_outs(struct hda_codec *codec, int num_pins, if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) name = "Bass Speaker"; + else if (num_pins == 2 && i == 1 && !strcmp(pfx, "Headphone")) + name = "Headphone2"; else if (num_pins >= 3) { snprintf(tmp, sizeof(tmp), "%s %s", pfx, channel_name[i]); -- 2.17.1

2 4

[alsa-devel] [RFC PATCH] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 06 Mar '19

06 Mar '19

the dpcm get from fe_clients/be_clients may be free before use possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 21 ++++++++++++++++++--- 3 files changed, 21 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db605955b..1e2be35ed36f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764dba147..d88757659729 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e82dea4..ac3f773ca768 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,9 +1294,11 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + spin_lock(&fe->card->dpcm_lock); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); + spin_unlock(&fe->card->dpcm_lock); } } @@ -1548,9 +1550,11 @@ void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock(&fe->card->dpcm_lock); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2640,11 +2644,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock(&fe->card->dpcm_lock); return ret; } @@ -3221,6 +3227,7 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,9 +3236,12 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) + state == SND_SOC_DPCM_STATE_SUSPEND) { + spin_unlock(&fe->card->dpcm_lock); return 0; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to free/stop this BE DAI */ return 1; @@ -3248,6 +3258,7 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,9 +3268,12 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) + state == SND_SOC_DPCM_STATE_PREPARE) { + spin_unlock(&fe->card->dpcm_lock); return 0; + } } + spin_unlock(&fe->card->dpcm_lock); /* it's safe to change hw_params */ return 1; @@ -3328,6 +3342,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock(&fe->card->dpcm_lock); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3363,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock(&fe->card->dpcm_lock); out: return offset; } -- 2.18.0

2 1

[alsa-devel] [PATCH RESEND v5 0/3] Add NXP AUDMIX device and machine drivers
by Viorel Suman 04 Mar '19

04 Mar '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V4: 1. Removed "model" attribute from device driver DT bindings documentation as suggested by Nicolin. 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 | 50 ++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 578 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++ sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ 6 files changed, 1078 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

2 5

[alsa-devel] [PATCH] ASoC: AMD: Configure master codec on all playback/capture cases
by Agrawal, Akshu 04 Mar '19

04 Mar '19

In the system design da7219 is the master codec and clocks are generated by it. Bclk is to be generated at the required rate for other codecs used when da7219 is acting only as clock master. For this call hw_params of da7219 during playback/capture on non da7219 codecs. Being able to set bclk at lower rate 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 | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a5daad973ce5..ad327415290a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -45,6 +45,7 @@ #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +static struct snd_soc_dai *codec_dai; static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; extern int bt_uart_enable; @@ -53,8 +54,10 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component; + + codec_dai = rtd->codec_dai; + component = codec_dai->component; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -255,6 +258,16 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static static int cz_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* da7219 Codec is clock master so setup as per the needs */ + if (codec_dai->driver->ops->hw_params) + return codec_dai->driver->ops->hw_params(substream, params, + codec_dai); + return 0; +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -268,16 +281,19 @@ static const struct snd_soc_ops cz_da7219_cap_ops = { 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

4 4

[alsa-devel] [PATCH RFC 0/3] tda998x updates for DAI formats and bclk_ratio
by Russell King - ARM Linux admin 04 Mar '19

04 Mar '19

This series addresses two issues with TDA998x that have been identified: 1) Peter found that the I2S format was not being explicitly set, and retains its value from whatever was previously running on the platform. Work around this by implementing support for setting the I2S format from the DAI format, rather than merely defaulting the register back to its power-on value. 2) Sven found that TDA998x does not work on his Freescale platform, which always uses a 64·fs bitclock. The TDA998x driver was deriving this information from the sample width, which, while it works for Beagle Bone Black, does not allow the driver to be used with other I2S sources that may have different behaviours. To work around that, we implement support for snd_soc_dai_set_bclk_ratio() in hdmi-codec, and propagate its value to TDA998x and other HDMI codecs via a new member. However, since snd_soc_dai_set_bclk_ratio() is never called, we need to avoid breaking any existing users, so we detect the lack of call by an impossible zero value, and subsitute a value corresponding with the TDA998x's old behaviour. It is hoped that snd_soc_dai_set_bclk_ratio() will see more adoption in ASoC, and the TDA998x specific defaulting can be removed. drivers/gpu/drm/i2c/tda998x_drv.c | 75 ++++++++++++++++++++++++++------------- include/drm/i2c/tda998x.h | 12 +++++-- include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 45 +++++++++++++++++++++-- 4 files changed, 104 insertions(+), 29 deletions(-) -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

5 19

[alsa-devel] [PATCH] ASoC: hdmi-codec: avoid limiting params->msbits in hw_params()
by Russell King 04 Mar '19

04 Mar '19

Limiting the value of the passed in params->msbits in the hw_params() callback is redundant on three counts: 1. We already specify in the DAI driver that we can only handle up to 24 bits. This means msbits will be limited to 24 via the ALSA constraints imposed by the ASoC core, unless we have multiple codecs that can handle more bits. 2. Nothing in our hw_params() implementation uses this value. 3. The copy of the params that we are passed by the ASoC core never reads back the msbits value. Consequently, this code is unnecessary and does nothing useful. Remove it. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d5f73c837281..35df73e42cbc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -484,9 +484,6 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, params_width(params), params_rate(params), params_channels(params)); - if (params_width(params) > 24) - params->msbits = 24; - ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, sizeof(hp.iec.status)); if (ret < 0) { -- 2.7.4

3 2

[alsa-devel] Regression: ASoC: soc-core: clear platform pointers on error
by Jon Hunter 04 Mar '19

04 Mar '19

Hi Mark, Curtis, I am seeing a regression on -next where the soundcard on one of our Tegra boards fails to initialise following a probe deferral. The bisect points to the commit 78a24e10cd94420f1b4e2dc5923ae7109e2aaba1 ('ASoC: soc-core: clear platform pointers on error') and reverting this on top of -next fixes the problem. Looking at the bootlog from the failure I see ... tegra-snd-wm8903 sound: ASoC: failed to init link WM8903 tegra-snd-wm8903 sound: snd_soc_register_card failed (-517) tegra30-i2s 70080400.i2s: DMA channels sourced from device 70080000.ahub tegra-snd-wm8903 sound: ASoC: Both platform name/of_node are set for WM8903 tegra-snd-wm8903 sound: ASoC: failed to init link WM8903 tegra-snd-wm8903 sound: snd_soc_register_card failed (-22) tegra-snd-wm8903: probe of sound failed with error -22 With the above change I see soc_cleanup_platform() is ever being called when the probe is deferred and hence leads to the failure. Note that the initial failure, "ASoC: failed to init link WM8903" occurs very early in snd_soc_register_card() when initialising the prelinks. The following fixes it, but I have not scrutinised the code to see if there are other exit points that we need to handle. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..6f66beb0c3ae 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); Cheers Jon -- nvpublic

2 2

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: fix reset gpio reference counting
by Philipp Puschmann 04 Mar '19

04 Mar '19

This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> --- Changes in v2: Fix typo in title tlv320aix3x -> tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

2 1

[alsa-devel] Applied "ASoC: tlv320aic3x: fix reset gpio reference counting" to the asoc tree
by Mark Brown 04 Mar '19

04 Mar '19

The patch ASoC: tlv320aic3x: fix reset gpio reference counting 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 82ad759143ed77673db0d93d53c1cde7b99917ee Mon Sep 17 00:00:00 2001 From: Philipp Puschmann <philipp.puschmann(a)emlix.com> Date: Wed, 27 Feb 2019 16:17:33 +0100 Subject: [PATCH] ASoC: tlv320aic3x: fix reset gpio reference counting This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: mediatek: btcvsd add loopback
by KaiChieh Chuang 04 Mar '19

04 Mar '19

add direct loopback path from rx to tx Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 69 +++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 1b8bcdaf..9a163d7 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -49,6 +49,7 @@ enum bt_sco_state { BT_SCO_STATE_IDLE, BT_SCO_STATE_RUNNING, BT_SCO_STATE_ENDING, + BT_SCO_STATE_LOOPBACK, }; enum bt_sco_direct { @@ -486,7 +487,8 @@ static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) if (bt->rx->state != BT_SCO_STATE_RUNNING && bt->rx->state != BT_SCO_STATE_ENDING && bt->tx->state != BT_SCO_STATE_RUNNING && - bt->tx->state != BT_SCO_STATE_ENDING) { + bt->tx->state != BT_SCO_STATE_ENDING && + bt->tx->state != BT_SCO_STATE_LOOPBACK) { dev_warn(bt->dev, "%s(), in idle state: rx->state: %d, tx->state: %d\n", __func__, bt->rx->state, bt->tx->state); goto irq_handler_exit; @@ -512,6 +514,42 @@ static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) buf_cnt_tx = btsco_packet_info[packet_type][2]; buf_cnt_rx = btsco_packet_info[packet_type][3]; + if (bt->tx->state == BT_SCO_STATE_LOOPBACK) { + u8 *src, *dst; + unsigned long connsys_addr_rx, ap_addr_rx; + unsigned long connsys_addr_tx, ap_addr_tx; + + connsys_addr_rx = *bt->bt_reg_pkt_r; + ap_addr_rx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_rx & 0xFFFF); + + connsys_addr_tx = *bt->bt_reg_pkt_w; + ap_addr_tx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_tx & 0xFFFF); + + if (connsys_addr_tx == 0xdeadfeed || + connsys_addr_rx == 0xdeadfeed) { + /* bt return 0xdeadfeed if read reg during bt sleep */ + dev_warn(bt->dev, "%s(), connsys_addr_tx == 0xdeadfeed\n", + __func__); + goto irq_handler_exit; + } + + src = (u8 *)ap_addr_rx; + dst = (u8 *)ap_addr_tx; + + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_BT2ARM, src, + bt->tx->temp_packet_buf, + packet_length, + packet_num); + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_ARM2BT, + bt->tx->temp_packet_buf, dst, + packet_length, + packet_num); + bt->rx->rw_cnt++; + bt->tx->rw_cnt++; + } + if (bt->rx->state == BT_SCO_STATE_RUNNING || bt->rx->state == BT_SCO_STATE_ENDING) { if (bt->rx->xrun) { @@ -1067,6 +1105,33 @@ static int btcvsd_band_set(struct snd_kcontrol *kcontrol, return 0; } +static int btcvsd_loopback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + bool lpbk_en = bt->tx->state == BT_SCO_STATE_LOOPBACK; + + ucontrol->value.integer.value[0] = lpbk_en; + return 0; +} + +static int btcvsd_loopback_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (ucontrol->value.integer.value[0]) { + mtk_btcvsd_snd_set_state(bt, bt->tx, BT_SCO_STATE_LOOPBACK); + mtk_btcvsd_snd_set_state(bt, bt->rx, BT_SCO_STATE_LOOPBACK); + } else { + mtk_btcvsd_snd_set_state(bt, bt->tx, BT_SCO_STATE_RUNNING); + mtk_btcvsd_snd_set_state(bt, bt->rx, BT_SCO_STATE_RUNNING); + } + return 0; +} + static int btcvsd_tx_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1202,6 +1267,8 @@ static int btcvsd_tx_timestamp_get(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new mtk_btcvsd_snd_controls[] = { SOC_ENUM_EXT("BTCVSD Band", btcvsd_enum[0], btcvsd_band_get, btcvsd_band_set), + SOC_SINGLE_BOOL_EXT("BTCVSD Loopback Switch", 0, + btcvsd_loopback_get, btcvsd_loopback_set), SOC_SINGLE_BOOL_EXT("BTCVSD Tx Mute Switch", 0, btcvsd_tx_mute_get, btcvsd_tx_mute_set), SOC_SINGLE_BOOL_EXT("BTCVSD Tx Irq Received Switch", 0, -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix S/PDIF DAI
by Russell King 04 Mar '19

04 Mar '19

When using the S/PDIF DAI, there is no requirement to call snd_soc_dai_set_fmt() as there is no DAI format definition that defines S/PDIF. In any case, S/PDIF does not have separate clocks, this is embedded into the data stream. Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt to configure TDA998x via the hw_params callback fails as the hdmi_codec_daifmt is left initialised to zero. Since the S/PDIF DAI will only be used by S/PDIF, prepare the hdmi_codec_daifmt structure for this format. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 118 +++++++++++++++++++++--------------------- 1 file changed, 59 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e5b6769b9797..d5f73c837281 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -529,73 +529,71 @@ static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); struct hdmi_codec_daifmt cf = { 0 }; - int ret = 0; dev_dbg(dai->dev, "%s()\n", __func__); - if (dai->id == DAI_ID_SPDIF) { - cf.fmt = HDMI_SPDIF; - } else { - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf.bit_clk_master = 1; - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFM: - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBM_CFS: - cf.bit_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } + if (dai->id == DAI_ID_SPDIF) + return 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - cf.frame_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - cf.bit_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_IF: - cf.frame_clk_inv = 1; - cf.bit_clk_inv = 1; - break; - } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - cf.fmt = HDMI_I2S; - break; - case SND_SOC_DAIFMT_DSP_A: - cf.fmt = HDMI_DSP_A; - break; - case SND_SOC_DAIFMT_DSP_B: - cf.fmt = HDMI_DSP_B; - break; - case SND_SOC_DAIFMT_RIGHT_J: - cf.fmt = HDMI_RIGHT_J; - break; - case SND_SOC_DAIFMT_LEFT_J: - cf.fmt = HDMI_LEFT_J; - break; - case SND_SOC_DAIFMT_AC97: - cf.fmt = HDMI_AC97; - break; - default: - dev_err(dai->dev, "Invalid DAI interface format\n"); - return -EINVAL; - } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; } hcp->daifmt[dai->id] = cf; - return ret; + return 0; } static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) @@ -792,8 +790,10 @@ static int hdmi_codec_probe(struct platform_device *pdev) i++; } - if (hcd->spdif) + if (hcd->spdif) { hcp->daidrv[i] = hdmi_spdif_dai; + hcp->daifmt[DAI_ID_SPDIF].fmt = HDMI_SPDIF; + } dev_set_drvdata(dev, hcp); -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: ab8500: Mark expected switch fall-through
by Gustavo A. R. Silva 04 Mar '19

04 Mar '19

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. This patch fixes the following warning: In file included from sound/soc/codecs/ab8500-codec.c:24: sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’: ./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=] _dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’ dev_err(dai->component->dev, ^~~~~~~ sound/soc/codecs/ab8500-codec.c:2132:2: note: here default: ^~~~~~~ Warning level 3 was used: -Wimplicit-fallthrough=3 This patch is part of the ongoing efforts to enable -Wimplicit-fallthrough. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/ab8500-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 03bbbcd3b6c1..87616b126018 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2129,6 +2129,7 @@ static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_err(dai->component->dev, "%s: ERROR: The device is either a master or a slave.\n", __func__); + /* fall through */ default: dev_err(dai->component->dev, "%s: ERROR: Unsupporter master mask 0x%x\n", -- 2.21.0

2 1

[alsa-devel] [PATCH 0/5] ASoC: stm32: sai: miscellaneous fixes
by Olivier Moysan 04 Mar '19

04 Mar '19

This patch-set is a collection of fixes for STM32 SAI driver. Olivier Moysan (5): ASoC: stm32: sai: fix iec958 controls indexation ASoC: stm32: sai: fix exposed capabilities in spdif mode ASoC: stm32: sai: fix race condition in irq handler ASoC: stm32: sai: fix oversampling mode ASoC: stm32: sai: fix set_sync service sound/soc/stm/stm32_sai.c | 7 ++++--- sound/soc/stm/stm32_sai_sub.c | 48 ++++++++++++++++++++++++++++++++++--------- 2 files changed, 42 insertions(+), 13 deletions(-) -- 2.7.4

2 10

[alsa-devel] [PATCH V2] ASoC: fsl_asrc: add constraint for the asrc of older version
by S.j. Wang 02 Mar '19

02 Mar '19

There is 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. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - switch to add constraint in startup function 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..31494f225037 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); + + /* channel_bits = 3 means older version on imx35*/ + 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 2

[alsa-devel] [PATCH] ASoC: cs42xx8: Remove S32_LE in format list
by S.j. Wang 02 Mar '19

02 Mar '19

cs42xx8 is a 24-bit A/D and 24-bit D/A device, so the S32_LE should not be in the supported format list. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..271972337d23 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -32,8 +32,7 @@ #define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) /* codec private data */ struct cs42xx8_priv { -- 1.9.1

4 5

[alsa-devel] [PATCH 0/3] ASoC: core/dapm/topology: minor fixes
by Pierre-Louis Bossart 02 Mar '19

02 Mar '19

3 small patches that have been used for a while but not contributed upstream. The first patch fixes a name mismatch when using the same machine driver on ApolloLake and GeminiLake Chromebooks. The second one fixes a minor mistake in the location of a topology pointer update, and aligns the code on the same pattern. The last one corrects yet another miss with the introduction of the new buffer/src/scheduler soc_dapm elements which were always showing as 'On' Liam Girdwood (1): ASoC: topology: Align tplg pointer increment across all kcontrols Pierre-Louis Bossart (1): ASoC: core: support driver alias names for FE topology overrides Ranjani Sridharan (1): ASoC: dapm: set power_check callback for widgets that shouldnt be always on sound/soc/soc-core.c | 7 +++++-- sound/soc/soc-dapm.c | 7 +++++++ sound/soc/soc-topology.c | 17 +++++++++-------- 3 files changed, 21 insertions(+), 10 deletions(-) -- 2.17.1

1 3

[alsa-devel] [PATCH 3/3] arm64: dts: hi3660-hikey960: Add i2s & sound device
by Pengcheng Li 01 Mar '19

01 Mar '19

From: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Tanglei Han <hantanglei(a)huawei.com> Signed-off-by: Guangke Ji <jiguangke(a)huawei.com> Signed-off-by: Feng Chen <puck.chen(a)hisilicon.com> Signed-off-by: Kaihua Zhong <zhongkaihua(a)huawei.com> Signed-off-by: Jun Chen <chenjun14(a)huawei.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Yiping Xu <xuyiping(a)hisilicon.com> Signed-off-by: Pengcheng Li <lipengcheng8(a)huawei.com> Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> --- arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts | 29 +++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts index 186251f..457cebc 100644 --- a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts +++ b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts @@ -27,6 +27,35 @@ /* rewrite this at bootloader */ reg = <0x0 0x0 0x0 0x0>; }; + + i2s2: hi3660_i2s { + compatible = "hisilicon,hi3660-i2s-1.0"; + reg = <0x0 0xe804f800 0x0 0x400>, + <0x0 0xe804e000 0x0 0x400>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pmx_func &i2s2_cfg_func>; + dmas = <&asp_dmac 18 &asp_dmac 19>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "hikey-hdmi"; + simple-audio-card,format = "i2s"; + + simple-audio-card,bitclock-master = <&sound_master>; + simple-audio-card,frame-master = <&sound_master>; + + sound_master: simple-audio-card,cpu { + sound-dai = <&i2s2>; + }; + + simple-audio-card,codec { + sound-dai = <&adv7533>; + }; + }; + }; &uart5 { -- 2.8.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_asrc: add protection for the asrc of older version
by S.j. Wang 01 Mar '19

01 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 protection when the asrc of older version is used. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 528e8b108422..b3b3c5e15ef1 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -109,7 +109,8 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) if (index == ASRC_INVALID_PAIR) { dev_err(dev, "all pairs are busy now\n"); ret = -EBUSY; - } else if (asrc_priv->channel_avail < channels) { + } else if (asrc_priv->channel_avail < channels || + (asrc_priv->channel_bits < 4 && channels % 2 != 0)) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; } else { -- 1.9.1

3 5

Re: [alsa-devel] [PATCH] ASoC: cs42xx8: Remove S32_LE in format list
by S.j. Wang 01 Mar '19

01 Mar '19

Hi > > cs42xx8 is a 24-bit A/D and 24-bit D/A device, so the S32_LE should > > not be in the supported format list. > > > > Signed-off-by: Shengjiu Wang<shengjiu.wang(a)nxp.com> > > Is the device capable of accepting 32-bit samples, even if it downgrades it to > 24-bit internally? If so, then maybe SNDRV_PCM_FMTBIT_S32_LE should > stay. Yes, the S32_LE can be accepted by device, it seems like that the device only Use the upper 24bit, so the S32_LE result is same as S24_LE, do you think the S32_LE should be remained? Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MOTU MicroBook II
by Manuel Reinhardt 28 Feb '19

28 Feb '19

Add an entry to the quirks-table to for usb-audio to recognize the Microbook II (although it only exposes vendor interfaces). A simple boot quirk is also implemented to set up the sample rate and make sure that no audio urbs are sent before the device is ready. This patch only provides audio playback and capture at 96kHz sample rate. Notice the following shortcomings: - The sample rate is currently hardcoded to 96k although the device also supports 48k and 44.1k. - The various mixer controls of the MicroBook are not made available. - The keep-iface control should be on by default because the device shuts down whenever the altsetting is reset which is usually unwanted. (I don't know the best way to do this) - The communication format used by the MicroBook for sample rate setting and also other setup has been reverse engineered by looking at the usbmon output while running the windows driver through virtualbox. In this patch the first byte of every message is set to \0 while in the observed communications the first byte acts as a "message-counter" increasing its value with every message sent. Leaving it at \0 does not seem to affect the device. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 4 ++ sound/usb/quirks-table.h | 65 +++++++++++++++++++++++++ sound/usb/quirks.c | 101 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 170 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index db114f3977e0..056af0a57b22 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + ep = 0x84; + ifnum = 0; + goto add_sync_ep_from_ifnum; } if (attr == USB_ENDPOINT_SYNC_ASYNC && diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b345beb447bd..86e80916a029 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3398,5 +3398,70 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* MOTU Microbook II */ +{ + USB_DEVICE(0x07fd, 0x0004), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MOTU", + .product_name = "MicroBookII", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x84, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x00d8, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x03, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x0120, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f372c624bbf4..e6ce1bbe6ca6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1000,6 +1000,105 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } + +#define MICROBOOK_BUF_SIZE 128 + +static int snd_usb_motu_microbookii_communicate(struct usb_device *dev, u8 *buf, + int buf_size, int *length) +{ + int err, actual_length; + + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x01), buf, *length, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "MicroBookII snd: ", DUMP_PREFIX_NONE, 16, 1, + buf, actual_length, false); + + memset(buf, 0, buf_size); + + err = usb_interrupt_msg(dev, usb_rcvintpipe(dev, 0x82), buf, buf_size, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "MicroBookII rcv: ", DUMP_PREFIX_NONE, 16, 1, + buf, actual_length, false); + + *length = actual_length; + return 0; +} + +static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) +{ + int err, actual_length, poll_attempts = 0; + static const u8 set_samplerate_seq[] = { 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x14, + 0x00, 0x00, 0x00, 0x01 }; + static const u8 poll_ready_seq[] = { 0x00, 0x04, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x18 }; + u8 *buf = kzalloc(MICROBOOK_BUF_SIZE, GFP_KERNEL); + + if (!buf) + return -ENOMEM; + + dev_info(&dev->dev, "Waiting for MOTU Microbook II to boot up...\n"); + + /* First we tell the device which sample rate to use. */ + memcpy(buf, set_samplerate_seq, sizeof(set_samplerate_seq)); + actual_length = sizeof(set_samplerate_seq); + err = snd_usb_motu_microbookii_communicate(dev, buf, MICROBOOK_BUF_SIZE, + &actual_length); + + if (err < 0) { + dev_err(&dev->dev, + "failed setting the sample rate for Motu MicroBook II: %d\n", + err); + goto free_buf; + } + + /* Then we poll every 100 ms until the device informs of its readiness. */ + while (true) { + if (++poll_attempts > 100) { + dev_err(&dev->dev, + "failed booting Motu MicroBook II: timeout\n"); + err = -ENODEV; + goto free_buf; + } + + memset(buf, 0, MICROBOOK_BUF_SIZE); + memcpy(buf, poll_ready_seq, sizeof(poll_ready_seq)); + + actual_length = sizeof(poll_ready_seq); + err = snd_usb_motu_microbookii_communicate( + dev, buf, MICROBOOK_BUF_SIZE, &actual_length); + if (err < 0) { + dev_err(&dev->dev, + "failed booting Motu MicroBook II: communication error %d\n", + err); + goto free_buf; + } + + /* the device signals its readiness through a message of the + * form + * XX 06 00 00 00 00 0b 18 00 00 00 01 + * If the device is not yet ready to accept audio data, the + * last byte of that sequence is 00. + */ + if (actual_length == 12 && buf[actual_length - 1] == 1) + break; + + msleep(100); + } + + dev_info(&dev->dev, "MOTU MicroBook II ready\n"); + +free_buf: + kfree(buf); + return err; +} + /* * Setup quirks */ @@ -1177,6 +1276,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_gamecon780_boot_quirk(dev); case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + return snd_usb_motu_microbookii_boot_quirk(dev); } return 0; -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MOTU MicroBook II
by Manuel Reinhardt 28 Feb '19

28 Feb '19

Add an entry to the quirks-table to for usb-audio to recognize the Microbook II (although it only exposes vendor interfaces). A simple boot quirk is also implemented to set up the sample rate and make sure that no audio urbs are sent before the device is ready. This patch only provides audio playback and capture at 96kHz sample rate. Notice the following shortcomings: - The sample rate is currently hardcoded to 96k although the device also supports 48k and 44.1k. - The various mixer controls of the MicroBook are not made available. - The keep-iface control should be on by default because the device shuts down whenever the altsetting is reset which is usually unwanted. (I don't know the best way to do this) - The communication format used by the MicroBook for sample rate setting and also other setup has been reverse engineered by looking at the usbmon output while running the windows driver through virtualbox. In this patch the first byte of every message is set to \0 while in the observed communications the first byte acts as a "message-counter" increasing its value with every message sent. Leaving it at \0 does not seem to affect the device. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 4 ++ sound/usb/quirks-table.h | 65 +++++++++++++++++++++++++++ sound/usb/quirks.c | 97 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 166 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index db114f3977e0..056af0a57b22 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + ep = 0x84; + ifnum = 0; + goto add_sync_ep_from_ifnum; } if (attr == USB_ENDPOINT_SYNC_ASYNC && diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b345beb447bd..86e80916a029 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3398,5 +3398,70 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* MOTU Microbook II */ +{ + USB_DEVICE(0x07fd, 0x0004), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MOTU", + .product_name = "MicroBookII", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x84, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x00d8, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x03, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x0120, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f372c624bbf4..c86a06dd97dd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1000,6 +1000,101 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } + +#define MICROBOOK_BUF_SIZE 128 + +static int snd_usb_motu_microbookii_communicate(struct usb_device *dev, u8 *buf, + int buf_size, int *length) +{ + int err, actual_length; + + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x01), buf, *length, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "snd: ", DUMP_PREFIX_NONE, 16, 1, buf, + actual_length, false); + + memset(buf, 0, buf_size); + + err = usb_interrupt_msg(dev, usb_rcvintpipe(dev, 0x82), buf, + buf_size, &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "rcv: ", DUMP_PREFIX_NONE, 16, 1, buf, + actual_length, false); + + *length = actual_length; + return 0; +} + +static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) +{ + int err, actual_length, poll_attempts = 0; + bool setup_finished = false; + static const u8 set_samplerate_seq[] = { 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x14, + 0x00, 0x00, 0x00, 0x01 }; + static const u8 poll_ready_seq[] = { 0x00, 0x04, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x18 }; + u8 *buf = kzalloc(MICROBOOK_BUF_SIZE, GFP_KERNEL); + + if (!buf) + return -ENOMEM; + + dev_info(&dev->dev, "Waiting for MOTU Microbook II to boot up...\n"); + + /* First we tell the device which sample rate to use. */ + memcpy(buf, set_samplerate_seq, ARRAY_SIZE(set_samplerate_seq)); + actual_length = ARRAY_SIZE(set_samplerate_seq); + err = snd_usb_motu_microbookii_communicate(dev, buf, MICROBOOK_BUF_SIZE, + &actual_length); + + if (err < 0) { + kfree(buf); + dev_err(&dev->dev, + "failed setting the sample rate for Motu MicroBook II: %d\n", + err); + return err; + } + + /* Then we poll every 100 ms until the device informs of its readiness. */ + while (!setup_finished) { + if (++poll_attempts > 100) { + kfree(buf); + dev_err(&dev->dev, + "failed booting Motu MicroBook II: timeout\n"); + return -ENODEV; + } + + memset(buf, 0, MICROBOOK_BUF_SIZE); + memcpy(buf, poll_ready_seq, ARRAY_SIZE(poll_ready_seq)); + + actual_length = ARRAY_SIZE(poll_ready_seq); + err = snd_usb_motu_microbookii_communicate( + dev, buf, MICROBOOK_BUF_SIZE, &actual_length); + if (err < 0) { + kfree(buf); + dev_err(&dev->dev, + "failed booting Motu MicroBook II: communication error\n"); + return err; + } + + if (buf[actual_length - 1] == 1) + setup_finished = true; + + msleep(100); + } + + kfree(buf); + + dev_info(&dev->dev, "MOTU MicroBook II ready\n"); + + return 0; +} + /* * Setup quirks */ @@ -1177,6 +1272,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_gamecon780_boot_quirk(dev); case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + return snd_usb_motu_microbookii_boot_quirk(dev); } return 0; -- 2.20.1

2 2

Re: [alsa-devel] [PATCH] dt-bindings: adau1977: Fix reset-gpios typo
by Rob Herring 28 Feb '19

28 Feb '19

On Fri, 15 Feb 2019 12:26:33 +0200, Bogdan Togorean wrote: > This change fixes a typo in the dt-binding examples (reset_gpio -> > reset-gpios). > Even though 'reset-gpio' is a valid construct for gpiolib the naming > 'reset-gpios' is more suited for dt-bindings documentation. > > Signed-off-by: Bogdan Togorean <bogdan.togorean(a)analog.com> > --- > Documentation/devicetree/bindings/sound/adi,adau1977.txt | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > Reviewed-by: Rob Herring <robh(a)kernel.org>

1 0

[alsa-devel] WM8510 on i.MX6
by Schrempf Frieder 28 Feb '19

28 Feb '19

Hi, I have a board with i.MX6 and a WM8510 codec and I can play audio using an old vendor kernel (3.14) and my own asoc machine driver [1]. Now I want to run this with something more recent (currently trying on 4.14) and upstream drivers like simple-card or fsl-asoc-card. The problem is I'm only getting a little bit of noise from the speakers but nothing more. I would really appreciate it if someone more knowledgeable could have a quick look and maybe come up with some ideas how I can proceed debugging. Thanks, Frieder [1] https://paste.ee/p/VuZbg My devicetree setup for the simple-card approach looks like this: sound { compatible = "simple-audio-card"; simple-audio-card,name = "wm8510-audio"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_codec>; simple-audio-card,frame-master = <&sound_codec>; simple-audio-card,widgets = "Speaker", "Ext Spk"; simple-audio-card,routing = "Ext Spk", "SPKOUTP", "Ext Spk", "SPKOUTN"; assigned-clocks = <&clks IMX6QDL_CLK_CKO2_SEL>, <&clks IMX6QDL_CLK_CKO2>; assigned-clock-parents = <&clks IMX6QDL_CLK_OSC>, <&clks IMX6QDL_CLK_CKO2_PODF>; assigned-clock-rates = <0 24000000>; sound_master: simple-audio-card,cpu { sound-dai = <&ssi2>; }; sound_codec: simple-audio-card,codec { sound-dai = <&codec>; clocks = <&clks IMX6QDL_CLK_CKO2>; }; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; ssi2 { fsl,audmux-port = <1>; fsl,port-config = < (IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSEL(4) | IMX_AUDMUX_V2_PTCR_TCSEL(4) | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR) IMX_AUDMUX_V2_PDCR_RXDSEL(4) >; }; aud5 { fsl,audmux-port = <4>; fsl,port-config = < IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(1) >; }; }; &ssi2 { status = "okay"; }; &i2c2 { status = "okay"; clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; codec: wm8510@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8510"; reg = <0x1a>; }; }; &iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sound_mclk>; pinctrl_sound_mclk: sound { /* WM8510 MCLK */ fsl,pins = < MX6QDL_PAD_NANDF_CS2__CCM_CLKO2 0x130b0 >; }; pinctrl_audmux: audmux { fsl,pins = < MX6QDL_PAD_KEY_COL0__AUD5_TXC 0x130b0 MX6QDL_PAD_KEY_COL1__AUD5_TXFS 0x130b0 MX6QDL_PAD_KEY_ROW1__AUD5_RXD 0x130b0 MX6QDL_PAD_KEY_ROW0__AUD5_TXD 0x130b0 >; }; };

2 2

[alsa-devel] Applied "ASoC: tlv320aic3x: fix reset gpio reference counting" to the asoc tree
by Mark Brown 28 Feb '19

28 Feb '19

The patch ASoC: tlv320aic3x: fix reset gpio reference counting 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 82ad759143ed77673db0d93d53c1cde7b99917ee Mon Sep 17 00:00:00 2001 From: Philipp Puschmann <philipp.puschmann(a)emlix.com> Date: Wed, 27 Feb 2019 16:17:33 +0100 Subject: [PATCH] ASoC: tlv320aic3x: fix reset gpio reference counting This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

1 0

[alsa-devel] hdmi-codec: modifying params in hw_params() callback
by Russell King - ARM Linux admin 28 Feb '19

28 Feb '19

Hi all, While looking at hdmi-codec issues, I spotted this code: static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { ... if (params_width(params) > 24) params->msbits = 24; params->msbits is a parameter that is negotiated and refined by libasound, and is part of the ALSA constraint system. The "Writing an ALSA driver" documentation says about the hw_params callback: "This is called when the hardware parameter (``hw_params``) is set up by the application, that is, once when the buffer size, the period size, the format, etc. are defined for the pcm substream." which suggests we should only be reading the parameters, not writing to them. What's more is that the msbits is a parameter that is refined with userspace, so surely the above should be a declared constraint? Digging a bit deeper, ASoC passes a private copy of the params to each codec - a copy is made, then fixups for TDM slots are applied, followed by any topology fixups by the DAI link (be_hw_params_fixup) before the codec driver's hw_params() callback is made. Afterwards, ASoC reads back the rate, channels and physical (memory) width and stores them in the codec's DAI structure. The msbits are not read. hdmi-codec also seems to do nothing with the msbits parameter other than the above code. So, all in all, it seems that the above code limiting msbits is redundant - nothing will read this modified value. Can we kill it? Thanks. -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

3 6

[alsa-devel] ftp.alsa-project.org connection problems
by Michael Niedermayer 28 Feb '19

28 Feb '19

Hi Iam not sure this is the correct place to report this to but i didnt see it being reported anywhere yet on the ML archive or google since a few days wget from ftp://ftp.alsa-project.org does not work anymore This was noticed as it broke the ffmpeg oss-fuzz build: https://oss-fuzz-build-logs.storage.googleapis.com/log-a252663b-b71c-49f3-8… and it is locally and on another server reproducable so it seems not a issue "on my side" intererstingly it works without pasv mode wget ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 --2019-02-25 18:30:15-- ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 => ‘alsa-lib-1.1.0.tar.bz2’ Resolving ftp.alsa-project.org (ftp.alsa-project.org)... 207.180.221.201 Connecting to ftp.alsa-project.org (ftp.alsa-project.org)|207.180.221.201|:21... connected. Logging in as anonymous ... Logged in! ==> SYST ... done. ==> PWD ... done. ==> TYPE I ... done. ==> CWD (1) /pub/lib ... done. ==> SIZE alsa-lib-1.1.0.tar.bz2 ... 929874 ==> PASV ... couldn't connect to 207.180.221.201 port 35333: No route to host wget --no-passive-ftp ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 --2019-02-25 18:30:22-- ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 => ‘alsa-lib-1.1.0.tar.bz2’ Resolving ftp.alsa-project.org (ftp.alsa-project.org)... 207.180.221.201 Connecting to ftp.alsa-project.org (ftp.alsa-project.org)|207.180.221.201|:21... connected. Logging in as anonymous ... Logged in! ==> SYST ... done. ==> PWD ... done. ==> TYPE I ... done. ==> CWD (1) /pub/lib ... done. ==> SIZE alsa-lib-1.1.0.tar.bz2 ... 929874 ==> PORT ... done. ==> RETR alsa-lib-1.1.0.tar.bz2 ... done. Length: 929874 (908K) (unauthoritative) 100%[===========================================================================================================================================================================================================================================================================>] 929,874 2.43MB/s in 0.4s 2019-02-25 18:30:22 (2.43 MB/s) - ‘alsa-lib-1.1.0.tar.bz2’ saved [929874] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB I know you won't believe me, but the highest form of Human Excellence is to question oneself and others. -- Socrates

2 4

[alsa-devel] [GIT PULL] ASoC updates for v5.1
by Mark Brown 28 Feb '19

28 Feb '19

The following changes since commit 00a399cad1a063e7665f06b6497a807db20441fd: ALSA: pcm: Revert capture stream behavior change in blocking mode (2019-02-08 16:54:31 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.1-2 for you to fetch changes up to 3146089d235b304e665dc551472ae9cb9ab58572: Merge branch 'asoc-5.1' into asoc-next (2019-02-26 16:22:47 +0000) ---------------------------------------------------------------- ASoC: More changes for v5.1 Another batch of changes for ASoC, no big core changes - it's mainly small fixes and improvements for individual drivers. - A big refresh and cleanup of the Samsung drivers, fixing a number of issues which allow the driver to be used with a wider range of userspaces. - Fixes for the Intel drivers to make them more standard so less likely to get bitten by core issues. - New driver for Cirrus Logic CS35L26. ---------------------------------------------------------------- Aaro Koskinen (3): ARM: OMAP: dts: N950/N9: fix onenand timings MIPS: OCTEON: don't set octeon_dma_bar_type if PCI is disabled staging: octeon: fix broken phylib usage Abel Vesa (1): clk: imx: Fix fractional clock set rate computation Adam Thomson (2): ASoC: da7219: Add support for master mode BCLK rate adjustment ASoC: da7219: Update TDM usage to be more flexible Aditya Swarup (1): drm/i915/cnl: Fix CNL macros for Voltage Swing programming Al Viro (2): btrfs: fix potential oops in device_list_add missing barriers in some of unix_sock ->addr and ->path accesses Alban Crequy (1): bpf, lpm: fix lookup bug in map_delete_elem Alex Deucher (4): drm/amdgpu: Add missing power attribute to APU check drm/radeon: check if device is root before getting pci speed caps drm/amdgpu/psp11: TA firmware is optional (v3) drm/amdgpu: Set DPM_FLAG_NEVER_SKIP when enabling PM-runtime Alexander Duyck (2): mm: Use fixed constant in page_frag_alloc instead of size + 1 net: Do not allocate page fragments that are not skb aligned Alexander Shiyan (1): gpu: ipu-v3: Fix i.MX51 CSI control registers offset Alexander Usyskin (1): mei: free read cb on ctrl_wr list flush Alexandre Belloni (2): clk: at91: fix at91sam9x5 peripheral clock number clk: at91: fix masterck name Alexandre Ghiti (1): riscv: Adjust mmap base address at a third of task size Alexandre Torgue (1): net: stmmac: handle endianness in dwmac4_get_timestamp Alexei Naberezhnov (1): md/raid5: fix 'out of memory' during raid cache recovery Alexei Starovoitov (5): Merge branch 'typedef-func_proto' bpf: run bpf programs with preemption disabled bpf: fix lockdep false positive in percpu_freelist bpf: fix potential deadlock in bpf_prog_register bpf: fix lockdep false positive in stackmap Alexey Brodkin (1): ARC: define ARCH_SLAB_MINALIGN = 8 Alexey Dobriyan (1): proc: fix /proc/net/* after setns(2) Alexey Kardashevskiy (1): powerpc/powernv/sriov: Register IOMMU groups for VFs Alexey Khoroshilov (3): net: stmmac: dwmac-rk: fix error handling in rk_gmac_powerup() usb: dwc3: exynos: Fix error handling of clk_prepare_enable net: mv643xx_eth: disable clk on error path in mv643xx_eth_shared_probe() Alistair Francis (1): RISC-V: defconfig: Enable Generic PCIE by default Amit Kucheria (2): thermal: of-thermal: Print name of device node with error thermal: cpu_cooling: Clarify error message Anders Roxell (3): netfilter: ipt_CLUSTERIP: fix warning unused variable cn usb: phy: fix link errors kasan: mark file common so ftrace doesn't trace it Andi Kleen (1): perf script: Fix crash with printing mixed trace point and other events Andre Przywara (1): clk: sunxi: A31: Fix wrong AHB gate number Andrea Arcangeli (1): mm/hugetlb.c: teach follow_hugetlb_page() to handle FOLL_NOWAIT Andrea Claudi (2): ipvs: fix dependency on nf_defrag_ipv6 ipvs: fix warning on unused variable Andreas Gruenbacher (1): gfs2: Revert "Fix loop in gfs2_rbm_find" Andreas Schwab (2): RISC-V: fix bad use of of_node_put tty/serial: use uart_console_write in the RISC-V SBL early console Andreas Ziegler (3): tracing: uprobes: Fix typo in pr_fmt string tracing/uprobes: Fix output for multiple string arguments tracing: probeevent: Correctly update remaining space in dynamic area Andrei Vagin (1): kernel/exit.c: release ptraced tasks before zap_pid_ns_processes Andrew Ford (1): ASoC: wm_adsp: Allow compressed buffers in any memory region Andrew Lunn (4): net: dsa: mv88e6xxx: Fix serdes irq setup going recursive gpio: vf610: Mask all GPIO interrupts net: dsa: mv88e6xxx: Fix counting of ATU violations MAINTAINERS: Update the ocores i2c bus driver maintainer, etc Andrew Morton (1): revert "initramfs: cleanup incomplete rootfs" Andrey Konovalov (11): kasan: fix assigning tags twice kasan, kmemleak: pass tagged pointers to kmemleak kmemleak: account for tagged pointers when calculating pointer range kasan, slub: move kasan_poison_slab hook before page_address kasan, slub: fix conflicts with CONFIG_SLAB_FREELIST_HARDENED kasan, slub: fix more conflicts with CONFIG_SLAB_FREELIST_HARDENED kasan: fix random seed generation for tag-based mode kasan: prevent tracing of tags.c kasan, slab: fix conflicts with CONFIG_HARDENED_USERCOPY kasan, slab: make freelist stored without tags kasan, slab: remove redundant kasan_slab_alloc hooks Andy Shevchenko (3): serial: 8250_pci: Make PCI class test non fatal dmaengine: dmatest: Abort test in case of mapping error gpiolib: acpi: Introduce ACPI_GPIO_QUIRK_ONLY_GPIOIO Aneesh Kumar K.V (3): powerpc/radix: Fix kernel crash with mremap() mtd: Use mtd->name when registering nvmem device mtd: powernv_flash: Fix device registration error Anoob Soman (1): scsi: libiscsi: Fix race between iscsi_xmit_task and iscsi_complete_task Anson Huang (1): ARM: dts: imx6sx: correct backward compatible of gpt Antony Pavlov (2): RISC-V: asm/page.h: fix spelling mistake "CONFIG_64BITS" -> "CONFIG_64BIT" RISC-V: Kconfig: fix spelling mistake "traget" -> "target" Ard Biesheuvel (3): arm64: kaslr: ensure randomized quantities are clean also when kaslr is off arm64, mm, efi: Account for GICv3 LPI tables in static memblock reserve table efi/arm: Revert "Defer persistent reservations until after paging_init()" Arnaldo Carvalho de Melo (5): perf python: Remove -fstack-clash-protection when building with some clang versions perf clang: Do not use 'return std::move(something)' tools headers uapi: Sync linux/in.h copy from the kernel sources perf symbols: Add fallback definitions for GELF_ST_VISIBILITY() perf trace: Support multiple "vfs_getname" probes Arnd Bergmann (20): Merge tag 'imx-fixes-5.0' of git://git.kernel.org/.../shawnguo/linux into arm/fixes Merge tag 'soc-fsl-fix-v5.0' of git://git.kernel.org/.../leo/linux into arm/fixes Merge tag 'imx-fixes-5.0-2' of git://git.kernel.org/.../shawnguo/linux into arm/fixes Merge tag 'pxa-fixes-5.0' of https://github.com/rjarzmik/linux into arm/fixes Merge tag 'renesas-fixes2-for-v5.0' of git://git.kernel.org/.../horms/renesas into arm/fixes Merge tag 'qcom-fixes-for-5.0-rc3' of git://git.kernel.org/.../agross/linux into arm/fixes Merge tag 'davinci-fixes-for-v5.0-part2' of git://git.kernel.org/.../nsekhar/linux-davinci into arm/fixes Merge tag 'omap-for-v5.0/fixes-rc3' of git://git.kernel.org/.../tmlind/linux-omap into arm/fixes Merge tag 'amlogic-fixes-2.1' of git://git.kernel.org/.../khilman/linux-amlogic into arm/fixes Merge tag 'sunxi-fixes-for-5.0' of git://git.kernel.org/.../sunxi/linux into arm/fixes Merge tag 'renesas-fixes3-for-v5.0' of git://git.kernel.org/.../horms/renesas into arm/fixes Merge tag 'omap-for-v5.0/fixes-rc4' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'v5.0-rockchip-dts32fixes-1' of git://git.kernel.org/.../mmind/linux-rockchip into arm/fixes Merge tag 'v5.0-rockchip-dts64fixes-1' of git://git.kernel.org/.../mmind/linux-rockchip into arm/fixes Merge tag 'imx-fixes-5.0-3' of git://git.kernel.org/.../shawnguo/linux into arm/fixes Merge tag 'omap-for-v5.0/fixes-rc5' of git://git.kernel.org/.../tmlind/linux-omap into arm/fixes Documentation: change linux-4.x references to 5.x phonet: fix building with clang Merge tag 'omap-for-v5.0/fixes-rc7-signed' of git://git.kernel.org/.../tmlind/linux-omap into arm/fixes Merge tag 'mvebu-fixes-5.0-2' of git://git.infradead.org/linux-mvebu into arm/fixes Artemy Kovalyov (1): RDMA/umem: Add missing initialization of owning_mm Arthur Demchenkov (1): ARM: dts: n900: fix mmc1 card detect gpio polarity Arthur Kiyanovski (2): net: ena: fix race between link up and device initalization net: ena: update driver version from 2.0.2 to 2.0.3 Arun Parameswaran (1): net: dsa: b53: Fix for failure when irq is not defined in dt Aurelien Aptel (1): CIFS: fix use-after-free of the lease keys Aviya Erenfeld (1): nl80211: Fix FTM per burst maximum value Axel Lin (1): gpio: altera-a10sr: Set proper output level for direction_output Aya Levin (1): net/mlx5e: Allow MAC invalidation while spoofchk is ON Bard liao (1): ASoC: topology: free created components in tplg load error Bart Van Assche (2): lib/test_rhashtable: Make test_insert_dup() allocate its hash table dynamically RDMA/srp: Rework SCSI device reset handling Bartosz Golaszewski (2): ARM: dts: da850: fix interrupt numbers for clocksource gpiolib: fix line event timestamps for nested irqs Baruch Siach (2): Revert "PCI: armada8k: Add support for gpio controlled reset signal" arm64: dts: clearfog-gt-8k: fix SGMII PHY reset signal Benedict Wong (1): xfrm: Make set-mark default behavior backward compatible Beniamino Galvani (1): qmi_wwan: apply SET_DTR quirk to Sierra WP7607 Benjamin Coddington (1): NFS: Don't use page_file_mapping after removing the page Bernard Pidoux (1): net/rose: fix NULL ax25_cb kernel panic Bert Kenward (1): sfc: initialise found bitmap in efx_ef10_mtd_probe Bhawanpreet Lakha (2): drm/amd/display: fix optimize_bandwidth func pointer for dce80 drm/amd/display: set clocks to 0 on suspend on dce80 Biju Das (1): ARM: dts: r8a7743: Convert to new LVDS DT bindings Bill Kuzeja (1): scsi: qla2xxx: Fix panic from use after free in qla2x00_async_tm_cmd Bin Liu (1): usb: phy: am335x: fix race condition in _probe Bjorn Andersson (1): pinctrl: qcom: qcs404: Correct SDC tile Bjorn Helgaas (1): net: Don't default Cavium PTP driver to 'y' Björn Töpel (3): xsk: do not remove umem from netdevice on fall-back to copy-mode Revert "xsk: simplify AF_XDP socket teardown" i40e: fix XDP_REDIRECT/XDP xmit ring cleanup race Bob Peterson (1): Revert "gfs2: read journal in large chunks to locate the head" Bob Tracy (1): tools uapi: fix Alpha support Bodong Wang (1): Revert "net/mlx5e: E-Switch, Initialize eswitch only if eswitch manager" Bogdan Togorean (3): ASoC: adau1977: Fix reset-gpios typo ASoC: adau1977: Add MICBIAS example in DT bindings ASoC: adau1977: Add support for setting MICBIAS via DT Boris Brezillon (4): mtd: spinand: Handle the case where PROGRAM LOAD does not reset the cache mtd: spinand: Fix the error/cleanup path in spinand_init() mtd: Remove a debug trace in mtdpart.c mtd: Make sure mtd->erasesize is valid even if the partition is of size 0 Borislav Petkov (2): MAINTAINERS: Add Andy and Darren as arch/x86/platform/ reviewers x86/a.out: Clear the dump structure initially Brian Foster (2): xfs: eof trim writeback mapping as soon as it is cached xfs: end sync buffer I/O properly on shutdown error Brian Norris (1): ath10k: correct bus type for WCN3990 Brian Welty (1): IB/{hfi1, qib}: Fix WC.byte_len calculation for UD_SEND_WITH_IMM Carlo Caione (2): arm: dts: meson: Fix IRQ trigger type for macirq arm64: dts: imx8mq: Fix boot from eMMC Catalin Marinas (1): arm64: Do not issue IPIs for user executable ptes Changbin Du (1): kprobe: Do not use uaccess functions to access kernel memory that can fault Chaotian Jing (1): mmc: mediatek: fix incorrect register setting of hs400_cmd_int_delay Charles Keepax (4): ASoC: wm_adsp: Factor out stripping padding from ADSP data ASoC: wm_adsp: Reorder some functions for improved clarity ASoC: wm_adsp: Refactor compress stream initialisation ASoC: wm_adsp: Improve logging messages Chen-Yu Tsai (7): ARM: dts: sun6i: Add clock-output-names to osc24M clock pinctrl: sunxi: Fix and simplify pin bank regulator handling pinctrl: sunxi: Consider pin_base when calculating regulator array index pinctrl: sunxi: Correct number of IRQ banks on H6 main pin controller mmc: sunxi: Disable HS-DDR mode for H5 eMMC controller by default mmc: sunxi: Filter out unsupported modes declared in the device tree ASoC: sunxi: sun50i-codec-analog: Rename hpvcc regulator supply to cpvdd Cheng-Yi Chiang (2): ASoC: qcom: Kconfig: select dmic for sdm845 ASoC: qcom: Kconfig: fix dependency for sdm845 Chengguang Xu (1): m68k: set proper major_num when specifying module param major_num Chris Brandt (1): serial: sh-sci: Do not free irqs that have already been freed Chris Wilson (2): drm/amdgpu: Transfer fences to dmabuf importer drm/i915/fbdev: Actually configure untiled displays Christian Brauner (4): selftests: add binderfs selftests binder: fix CONFIG_ANDROID_BINDER_DEVICES binderfs: respect limit on binder control creation binderfs: remove separate device_initcall() Christian König (1): drm/amdgpu: disable bulk moves for now Christoffer Dall (2): KVM: arm/arm64: Reset the VCPU without preemption and vcpu state loaded KVM: arm/arm64: vgic: Always initialize the group of private IRQs Christoph Hellwig (5): iomap: fix a use after free in iomap_dio_rw drm/vmwgfx: remove CONFIG_X86 ifdefs drm/vmwgfx: remove CONFIG_INTEL_IOMMU ifdefs v2 drm/vmwgfx: fix the check when to use dma_alloc_coherent drm/vmwgfx: unwind spaghetti code in vmw_dma_select_mode Chuck Lever (1): svcrdma: Remove max_sge check at connect time Clint Taylor (1): drm/i915/icl: combo port vswing programming changes per BSPEC Codrin Ciubotariu (8): dmaengine: at_xdmac: Fix wrongfull report of a channel as in use ASoC: codecs: ad193x: Remove capture support for codecs without ADC ASoC: codecs: ad193x: Set constraint to always have 32 sample bits ASoC: codecs: ad193x: Fix frame polarity for DSP_A format ASoC: codecs: ad193x: Add runtime support for DSP_A and I2S modes ASoC: codecs: ad193x: Add support to disable on-chip PLL ASoC: codecs: pcm186x: fix wrong usage of DECLARE_TLV_DB_SCALE() ASoC: codecs: pcm186x: Fix energysense SLEEP bit Colin Ian King (5): selftests: cpu-hotplug: fix case where CPUs offline > CPUs present x86/fault: Fix sign-extend unintended sign extension ieee802154: mcr20a: fix indentation, remove tabs ASoC: fsi: fix spelling mistake "doens't" -> "doesn't" net/mlx4_en: fix spelling mistake: "quiting" -> "quitting" Cong Wang (6): netrom: switch to sock timer API xfrm: destroy xfrm_state synchronously on net exit path net_sched: fix a race condition in tcindex_destroy() net_sched: fix a memory leak in cls_tcindex net_sched: fix two more memory leaks in cls_tcindex team: avoid complex list operations in team_nl_cmd_options_set() Corentin Labbe (2): xtensa: Fix typo use space=>user space xtensa: rename BUILTIN_DTB to BUILTIN_DTB_SOURCE Damien Le Moal (1): scsi: sd_zbc: Fix zone information messages Dan Carpenter (9): clk: ti: Fix error handling in ti_clk_parse_divider_data() scsi: bnx2fc: Fix error handling in probe() scsi: 53c700: pass correct "dev" to dma_alloc_attrs() skge: potential memory corruption in skge_get_regs() lib/test_kmod.c: potential double free in error handling net: dsa: Fix NULL checking in dsa_slave_set_eee() net: dsa: bcm_sf2: potential array overflow in bcm_sf2_sw_suspend() ASoC: dapm: Potential small memory leak in dapm_cnew_widget() ASoC: cs35l36: Fix an IS_ERR() vs NULL checking bug Dan Murphy (1): iio: ti-ads8688: Update buffer allocation for timestamps Daniel Baluta (1): ASoC: simple-card: Fix refcount underflow Daniel Borkmann (6): bpf, doc: add reviewers to maintainers entry ipvlan, l3mdev: fix broken l3s mode wrt local routes Merge branch 'bpf-lockdep-fixes' ipvlan: disallow userns cap_net_admin to change global mode/flags Merge branch 'bpf-nfp-codegen-fixes' bpf, doc: add bpf list as secondary entry to maintainers file Daniel Vetter (1): mm: don't let userspace spam allocations warnings Darren Hart (VMware) (1): Documentation/ABI: Correct mlxreg-io KernelVersion for 5.0 Darrick J. Wong (2): xfs: set buffer ops when repair probes for btree type tmpfs: fix link accounting when a tmpfile is linked in Dave Airlie (11): Merge branch 'drm-fixes-5.0' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge branch 'drm-fixes-5.0' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge branch 'vmwgfx-fixes-5.0-2' of git://people.freedesktop.org/~thomash/linux into drm-fixes Merge tag 'drm-intel-fixes-2019-02-07-1' of git://anongit.freedesktop.org/drm/drm-intel into drm-fixes Merge tag 'drm-misc-fixes-2019-02-07' of git://anongit.freedesktop.org/drm/drm-misc into drm-fixes Merge tag 'imx-drm-fixes-2019-02-12' of git://git.pengutronix.de/pza/linux into drm-fixes Merge tag 'drm-misc-fixes-2019-02-13' of git://anongit.freedesktop.org/drm/drm-misc into drm-fixes Merge tag 'drm-intel-fixes-2019-02-13' of git://anongit.freedesktop.org/drm/drm-intel into drm-fixes Merge branch 'drm-fixes-5.0' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge tag 'drm-intel-fixes-2019-02-20' of git://anongit.freedesktop.org/drm/drm-intel into drm-fixes Merge branch 'drm-fixes-5.0' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Dave Chinner (2): Revert "mm: don't reclaim inodes with many attached pages" Revert "mm: slowly shrink slabs with a relatively small number of objects" Dave Watson (2): net: tls: Save iv in tls_rec for async crypto requests net: tls: Fix deadlock in free_resources tx David Chen (2): r8152: Add support for MAC address pass through on RTL8153-BD r8152: Fix an error on RTL8153-BD MAC Address Passthrough support David Disseldorp (1): scsi: target: make the pi_prot_format ConfigFS path readable David Hildenbrand (1): mm: migrate: don't rely on __PageMovable() of newpage after unlocking it David Howells (3): assoc_array: Fix shortcut creation keys: Fix dependency loop between construction record and auth key keys: Timestamp new keys David S. Miller (49): Merge branch 'master' of git://git.kernel.org/.../klassert/ipsec Merge tag 'mlx5-fixes-2019-01-25' of git://git.kernel.org/.../saeed/linux Merge branch 'hns-fixes' Merge git://git.kernel.org/.../pablo/nf Merge branch 'qed-Bug-fixes' Merge branch 'net-various-compat-ioctl-fixes' Merge branch 'erspan-always-reports-output-key-to-userspace' Merge branch 'virtio_net-Fix-problems-around-XDP-tx-and-napi_tx' Merge branch 'stmmac-fixes' Merge branch 'ieee802154-for-davem-2019-01-31' of git://git.kernel.org/.../sschmidt/wpan Merge tag 'mac80211-for-davem-2019-02-01' of git://git.kernel.org/.../jberg/mac80211 Merge tag 'batadv-net-for-davem-20190201' of git://git.open-mesh.org/linux-merge Merge branch 'smc-fixes' Merge git://git.kernel.org/.../bpf/bpf Merge branch 'vsock-virtio-hot-unplug' Merge branch 'smc-fixes' Merge tag 'wireless-drivers-for-davem-2019-02-04' of git://git.kernel.org/.../kvalo/wireless-drivers Merge branch 's390-qeth-fixes' Merge git://git.kernel.org/.../pablo/nf Merge tag 'mlx5-fixes-2019-02-05' of git://git.kernel.org/.../saeed/linux Merge branch 'qed-Bug-fixes' Merge branch 'ipv6-fixes' Merge branch 'r8169-revert-two-commits-due-to-a-regression' Merge git://git.kernel.org/.../pablo/nf Merge tag 'mac80211-for-davem-2019-02-12' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'ena-races' Merge branch 'net_sched-some-fixes-for-cls_tcindex' Merge tag 'mlx5-fixes-2019-02-13' of git://git.kernel.org/.../saeed/linux Merge git://git.kernel.org/.../pablo/nf Merge branch 'net-phy-fix-locking-issue' Merge tag 'mac80211-for-davem-2019-02-15' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'net-dsa-b53-VLAN-and-L2-fixes' net: Add header for usage of fls64() Merge git://git.kernel.org/.../bpf/bpf Merge branch 'tcp-fix-possible-crash-in-tcp_v4_err' Merge branch 'netdev-page_frag_alloc-fixes' Merge branch 'qed-iWARP' Merge tag 'wireless-drivers-for-davem-2019-02-18' of git://git.kernel.org/.../kvalo/wireless-drivers Merge git://git.kernel.org/.../pablo/nf Merge branch 'ipv6-route-rcu' Merge branch '40GbE' of git://git.kernel.org/.../jkirsher/net-queue Merge branch 'tipc-improvement-for-wait-and-wakeup' Merge branch 'report-erspan-version-field-just-for-erspan-tunnels' Merge branch 'master' of git://git.kernel.org/.../klassert/ipsec Merge branch 'nic-thunderx-fix-communication-races-between-VF-PF' Merge tag 'mac80211-for-davem-2019-02-22' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'bnxt_en-firmware-message-delay-fixes' Merge branch 'udp-a-few-fixes' Merge git://git.kernel.org/.../bpf/bpf David Sterba (1): btrfs: clean up pending block groups when transaction commit aborts Davidlohr Bueso (2): futex: Fix barrier comment xsk: share the mmap_sem for page pinning Deepa Dinamani (1): selftests: fix timestamping Makefile Deepak Rawat (1): drm/vmwgfx: Also check for crtc status while checking for DU active Derek Basehore (1): clk: Remove global clk traversal on fetch parent index Dietmar Eggemann (1): ARM: 8824/1: fix a migrating irq bug when hotplug cpu Dmitry Torokhov (7): pinctrl: cherryview: fix Strago DMI workaround Input: cap11xx - switch to using set_brightness_blocking() Input: ps2-gpio - flush TX work when closing port Input: matrix_keypad - use flush_delayed_work() Input: qt2160 - switch to using brightness_set_blocking() Revert "Input: elan_i2c - add ACPI ID for touchpad in ASUS Aspire F5-573G" Input: apanel - switch to using brightness_set_blocking() Dmitry V. Levin (1): parisc: Fix ptrace syscall number modification Dmitry Voytik (1): arm64: dts: rockchip: enable usb-host regulators at boot on rk3328-rock64 Doug Smythies (1): cpuidle: poll_state: Fix default time limit Douglas Gilbert (1): scsi: scsi_debug: fix write_same with virtual_gb problem Eli Cooper (1): netfilter: ipv6: Don't preserve original oif for loopback address Enric Balletbo i Serra (1): arm64: dts: rockchip: fix graph_port warning on rk3399 bob kevin and excavator Eric Anholt (1): drm/sched: Always trace the dependencies we wait on, to fix a race. Eric Biggers (3): KEYS: allow reaching the keys quotas exactly KEYS: user: Align the payload buffer KEYS: always initialize keyring_index_key::desc_len Eric Dumazet (12): rds: fix refcount bug in rds_sock_addref dccp: fool proof ccid_hc_[rt]x_parse_options() mISDN: fix a race in dev_expire_timer() rxrpc: bad unlock balance in rxrpc_recvmsg vxlan: test dev->flags & IFF_UP before calling netif_rx() net/x25: do not hold the cpu too long in x25_new_lci() batman-adv: fix uninit-value in batadv_interface_tx() net: fix possible overflow in __sk_mem_raise_allocated() tcp: clear icsk_backoff in tcp_write_queue_purge() tcp: tcp_v4_err() should be more careful net/x25: fix a race in x25_bind() tcp: repaired skbs must init their tso_segs Eric W. Biederman (6): signal: Make siginmask safe when passed a signal of 0 btrfs: On error always free subvol_name in btrfs_mount signal: Always attempt to allocate siginfo for SIGSTOP signal: Always notice exiting tasks signal: Better detection of synchronous signals signal: Restore the stop PTRACE_EVENT_EXIT Erik Schmauss (1): ACPI: Set debug output flags independent of ACPICA Eugene Loh (1): kallsyms: Handle too long symbols in kallsyms.c Eugeniy Paltsev (5): ARCv2: Enable unaligned access in early ASM code ARCv2: lib: memcpy: fix doing prefetchw outside of buffer ARC: fix actionpoints configuration detection ARC: U-boot: check arguments paranoidly ARC: enable uboot support unconditionally Fathi Boudra (2): selftests: net: use LDLIBS instead of LDFLAGS selftests: timers: use LDLIBS instead of LDFLAGS Felix Fietkau (3): batman-adv: release station info tidstats mac80211: ensure that mgmt tx skbs have tailroom for encryption mac80211: allocate tailroom for forwarded mesh packets Feras Daoud (1): IB/ipoib: Fix for use-after-free in ipoib_cm_tx_start Fernando Fernandez Mancera (1): netfilter: nfnetlink_osf: add missing fmatch check Filipe Manana (1): Btrfs: fix deadlock when allocating tree block during leaf/node split Florian Fainelli (7): net: systemport: Fix WoL with password after deep sleep net: dsa: b53: Fix default VLAN ID net: dsa: b53: Properly account for VLAN filtering net: systemport: Fix reception of BPDUs net: dsa: bcm_sf2: Do not assume DSA master supports WoL net: dsa: b53: Do not program CPU port's PVID Documentation: networking: switchdev: Update port parent ID section Florian Westphal (15): selftests: xfrm: add block rules with adjacent/overlapping subnets xfrm: policy: use hlist rcu variants on inexact insert, part 2 xfrm: policy: increment xfrm_hash_generation on hash rebuild xfrm: policy: delete inexact policies from inexact list on hash rebuild xfrm: policy: fix reinsertion on node merge selftests: xfrm: alter htresh to trigger move of policies to hash table xfrm: policy: fix infinite loop when merging src-nodes xfrm: refine validation of template and selector families netfilter: nft_compat: use refcnt_t type for nft_xt reference count netfilter: nft_compat: make lists per netns netfilter: nft_compat: destroy function must not have side effects netfilter: ebtables: compat: un-break 32bit setsockopt when no rules are present selftests: netfilter: add simple masq/redirect test cases netfilter: nft_compat: don't use refcount_inc on newly allocated entry netfilter: nat: fix spurious connection timeouts Francesco Ruggeri (1): netfilter: compat: initialize all fields in xt_init Gabriel Fernandez (1): Input: st-keyscan - fix potential zalloc NULL dereference Gao Xiang (1): staging: erofs: keep corrupted fs from crashing kernel in erofs_namei() Geert Uytterhoeven (3): arm64: dts: renesas: r8a774a1: Enable DMA for SCIF2 arm64: dts: renesas: r8a7796: Enable DMA for SCIF2 arm64: dts: renesas: r8a77965: Enable DMA for SCIF2 George Amanakis (1): tun: move the call to tun_set_real_num_queues George Wilkie (1): team: use operstate consistently for linkup Gerald Schaefer (1): iommu/vt-d: Fix memory leak in intel_iommu_put_resv_regions() Gilad Ben-Yossef (1): crypto: ccree - fix resume race condition on init Govindarajulu Varadarajan (1): enic: fix checksum validation for IPv6 Greg Kroah-Hartman (11): debugfs: fix debugfs_rename parameter checking debugfs: return error values, not NULL debugfs: debugfs_lookup() should return NULL if not found Revert "staging: erofs: keep corrupted fs from crashing kernel in erofs_namei()" Merge tag 'fixes-for-v5.0-rc4' of git://git.kernel.org/.../balbi/usb into usb-linus relay: check return of create_buf_file() properly blk-mq: protect debugfs_create_files() from failures serial: fix race between flush_to_ldisc and tty_open sctp: walk the list of asoc safely Merge tag 'iio-fixes-5.0a' of git://git.kernel.org/.../jic23/iio into staging-linus rpc: properly check debugfs dentry before using it Guennadi Liakhovetski (1): ASoC: regulator notifier registration should be managed Guenter Roeck (1): hwmon: (nct6775) Fix fan6 detection for NCT6793D Guido Günther (1): dt-bindings: imx8mq: Number clocks consecutively Guo Ren (6): riscv: fixup max_low_pfn with PFN_DOWN. csky: Fixup _PAGE_GLOBAL bit for 610 tlb entry csky: Fixup wrong pt_regs size csky: coding convention: Use task_stack_page csky: Fixup io-range page attribute for mmap("/dev/mem") csky: Fixup dead loop in show_stack Gustavo A. R. Silva (4): usb: gadget: udc: net2272: Fix bitwise and boolean operations drm/amd/powerplay: Fix missing break in switch tty: serial: 8250_mtk: Fix potential NULL pointer dereference perf tests evsel-tp-sched: Fix bitwise operator Guy Shattah (1): net/mlx5e: Use the inner headers to determine tc/pedit offload limitation on decap flows Hangbin Liu (4): geneve: should not call rt6_lookup() when ipv6 was disabled sit: check if IPv6 enabled before calling ip6_err_gen_icmpv6_unreach() net: vrf: remove MTU limits for vrf device Revert "bridge: do not add port to router list when receives query with source 0.0.0.0" Hans Wippel (1): net/smc: allow 16 byte pnetids in netlink policy Hans de Goede (2): iio: adc: axp288: Fix TS-pin handling libata: Add NOLPM quirk for SAMSUNG MZ7TE512HMHP-000L1 SSD Harald Freudenberger (1): s390/zcrypt: fix specification exception on z196 during ap probe Harini Katakam (1): net: macb: Apply RXUBR workaround only to versions with errata Hauke Mehrtens (2): net: Fix for_each_netdev_feature on Big endian net: dsa: Remove documentation for port_fdb_prepare He Kuang (1): perf top: Fix wrong hottest instruction highlighted Hedi Berriche (1): x86/platform/UV: Use efi_runtime_lock to serialise BIOS calls Heiko Schocher (1): ARM: dts: am335x-shc.dts: fix wrong cd pin level Heiner Kallweit (5): Revert "r8169: remove unneeded mmiowb barriers" Revert "r8169: make use of xmit_more and __netdev_sent_queue" net: phy: fix interrupt handling in non-started states net: phy: don't use locking in phy_is_started net: phy: fix potential race in the phylib state machine Helge Deller (1): CREDITS/MAINTAINERS: Retire parisc-linux.org email domain Herbert Xu (2): mac80211: Use linked list instead of rhashtable walk for mesh tables mac80211: Free mpath object when rhashtable insertion fails Heyi Guo (1): irqchip/gic-v4: Fix occasional VLPI drop Hoang Le (1): tipc: fix skb may be leaky in tipc_link_input Huacai Chen (2): cacheinfo: Keep the old value if of_property_read_u32 fails MIPS: Loongson: Introduce and use loongson_llsc_mb() Huang Rui (1): drm/amdgpu: fix the incorrect external id for raven series Huang Zijiang (1): net: hns: Fix object reference leaks in hns_dsaf_roce_reset() Hui Wang (1): ALSA: hda/realtek: Disable PC beep in passthrough on alc285 Huy Nguyen (1): net/mlx5: No command allowed when command interface is not ready Ian Kent (1): autofs: fix error return in autofs_fill_super() Ilan Peer (1): mac80211: Fix Tx aggregation session tear down with ITXQs Ilya Dryomov (1): libceph: handle an empty authorize reply Ingo Molnar (3): Merge tag 'perf-urgent-for-mingo-5.0-20190121' of git://git.kernel.org/.../acme/linux into perf/urgent Merge tag 'perf-urgent-for-mingo-5.0-20190205' of git://git.kernel.org/.../acme/linux into perf/urgent perf/core: Fix impossible ring-buffer sizes warning J. Bruce Fields (1): Revert "nfsd4: return default lease period" Jack Morgenstein (1): IB/mlx4: Fix using wrong function to destroy sqp AHs under SRIOV Jacob Wen (2): l2tp: fix reading optional fields of L2TPv3 l2tp: copy 4 more bytes to linear part if necessary Jakub Kicinski (1): tools: bpftool: fix crash with un-owned prog arrays Jakub Sitnicki (1): sk_msg: Always cancel strp work before freeing the psock James Bottomley (2): scsi: aic94xx: fix module loading scsi: sd: fix entropy gathering for most rotational disks James Morse (4): arm64: kprobe: Always blacklist the KVM world-switch code arm64: hyp-stub: Forbid kprobing of the hyp-stub arm64: hibernate: Clean the __hyp_text to PoC after resume KVM: arm64: Forbid kprobing of the VHE world-switch code James Schulman (2): ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier ASoC: cs35l36: Add device tree documentation for CS35L36 Jan Kara (3): mm: migrate: make buffer_migrate_page_norefs() actually succeed fs/drop_caches.c: avoid softlockups in drop_pagecache_sb() blktrace: Show requests without sector Jan Sokolowski (1): ixgbe: don't do any AF_XDP zero-copy transmit if netif is not OK Jani Nikula (2): drm/i915/opregion: fix version check drm/i915/opregion: rvda is relative from opregion base in opregion 2.1+ Jann Horn (6): fuse: call pipe_buf_release() under pipe lock kvm: fix kvm_ioctl_create_device() reference counting (CVE-2019-6974) netfilter: nf_nat_snmp_basic: add missing length checks in ASN.1 cbs mm: page_alloc: fix ref bias in page_frag_alloc() for 1-byte allocs MAINTAINERS: mark CAIF as orphan net: socket: add check for negative optlen in compat setsockopt Jason Gunthorpe (2): RDMA/device: Expose ib_device_try_get(() IB/mlx5: Fix how advise_mr() launches async work Jason Kridner (1): pinctrl: mcp23s08: spi: Fix regmap allocation for mcp23s18 Jason Wang (2): vhost: fix OOB in get_rx_bufs() vhost: correctly check the return value of translate_desc() in log_used() Jay Cornwall (1): drm/amdgpu: Implement doorbell self-ring for NBIO 7.4 Jean-Philippe Brucker (1): arm64: kexec_file: handle empty command-line Jeff Kirsher (1): ixgbe: fix older devices that do not support IXGBE_MRQC_L3L4TXSWEN Jenny TC (1): ASoC: Intel: Boards: Add Maxim98373 support Jens Axboe (5): Merge branch 'md-fixes' of https://github.com/liu-song-6/linux into for-linus ide: ensure atapi sense request aren't preempted Merge branch 'nvme-5.0' of git://git.infradead.org/nvme into for-linus Merge branch 'md-fixes' of https://github.com/liu-song-6/linux into for-linus Merge branch 'nvme-5.0' of git://git.infradead.org/nvme into for-linus Jeremy Soller (1): ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5 Jernej Skrabec (2): ARM: dts: sun8i: h3: Add ethernet0 alias to Beelink X2 arm64: dts: allwinner: a64: Fix USB OTG regulator Jerry Snitselaar (2): iommu/amd: Call free_iova_fast with pfn in map_sg iommu/amd: Unmap all mapped pages in error path of map_sg Jesper Dangaard Brouer (1): MAINTAINERS: Add entry for XDP (eXpress Data Path) Jiada Wang (1): ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address Jianchao Wang (2): blk-mq: fix a hung issue when fsync blk-mq: insert rq with DONTPREP to hctx dispatch list when requeue Jiong Wang (2): nfp: bpf: fix code-gen bug on BPF_ALU | BPF_XOR | BPF_K nfp: bpf: fix ALU32 high bits clearance bug Jiri Olsa (4): perf ordered_events: Fix crash in ordered_events__free bpftool: Fix prog dump by tag perf symbols: Filter out hidden symbols from labels perf/x86: Add check_period PMU callback Jisheng Zhang (2): i3c: fix missing detach if failed to retrieve i3c dev i3c: master: dw: fix deadlock Joerg Roedel (1): iommu/mediatek: Use correct fwspec in mtk_iommu_add_device() Johan Hovold (4): staging: speakup: fix tty-operation NULL derefs dt-bindings: sound: msm8916-wcd-analog: fix example regulator names ASoC: msm8916-wcd-analog: add missing license information ASoC: msm8916-wcd-digital: convert license header to SPDX Johan Jonker (1): ARM: dts: rockchip: remove qos_cif1 from rk3188 power-domain Johannes Berg (11): decnet: fix DN_IFREQ_SIZE Revert "socket: fix struct ifreq size in compat ioctl" Revert "kill dev_ifsioc()" net: socket: fix SIOCGIFNAME in compat net: socket: make bond ioctls go through compat_ifreq_ioctl() cfg80211: call disconnect_wk when AP stops cfg80211: pmsr: fix MAC address setting cfg80211: pmsr: fix abort locking mac80211: call drv_ibss_join() on restart cfg80211: pmsr: record netlink port ID cfg80211: prevent speculation on cfg80211_classify8021d() return Johannes Weiner (4): psi: fix aggregation idle shut-off psi: clarify the Kconfig text for the default-disable option x86/resctrl: Avoid confusion over the new X86_RESCTRL config psi: avoid divide-by-zero crash inside virtual machines John David Anglin (1): dsa: mv88e6xxx: Ensure all pending interrupts are handled prior to exit John Garry (1): scsi: libsas: Fix rphy phy_identifier for PHYs with end devices attached John Johansen (1): apparmor: Fix aa_label_build() error handling for failed merges Jonathan Bakker (2): Input: pwm-vibra - prevent unbalanced regulator Input: bma150 - register input device after setting private data Jonathan Neuschäfer (1): init/Kconfig: fix grammar by moving a closing parenthesis Joonas Lahtinen (1): drm/i915: Prevent a race during I915_GEM_MMAP ioctl with WC set Jose Abreu (4): net: stmmac: Fallback to Platform Data clock in Watchdog conversion net: stmmac: Send TSO packets always from Queue 0 net: stmmac: Disable EEE mode earlier in XMIT callback net: stmmac: Fix a race in EEE enable callback Josef Bacik (1): btrfs: don't end the transaction for delayed refs in throttle Josh Elsasser (1): net: set default network namespace in init_dummy_netdev() Josh Poimboeuf (1): cpu/hotplug: Fix "SMT disabled by BIOS" detection for KVM Jouke Witteveen (1): Documentation: bring operstate documentation up-to-date Juergen Gross (1): x86/mm: Make set_pmd_at() paravirt aware Julia Lawall (1): drm/imx: imx-ldb: add missing of_node_puts Julian Wiedmann (4): s390/qeth: release cmd buffer in error paths s390/qeth: fix use-after-free in error path s390/qeth: cancel close_dev work before removing a card s390/qeth: conclude all event processing before offlining a card Julien Thierry (3): KVM: arm/arm64: vgic: Make vgic_irq->irq_lock a raw_spinlock KVM: arm/arm64: vgic: Make vgic_dist->lpi_list_lock a raw_spinlock KVM: arm/arm64: vgic: Make vgic_cpu->ap_list_lock a raw_spinlock Jun-Ru Chang (1): MIPS: Remove function size check in get_frame_info() Kai-Heng Feng (1): sky2: Increase D3 delay again KaiChieh Chuang (2): ASoC: mediatek: use %pOFn instead of device_node.name ASoC: mediatek: btcvsd fix rx stream assign Kairui Song (1): x86/kexec: Don't setup EFI info if EFI runtime is not enabled Kal Conley (1): net/packet: fix 4gb buffer limit due to overflow check Kalash Nainwal (1): net: Set rtm_table to RT_TABLE_COMPAT for ipv6 for tables > 255 Kan Liang (2): x86/cpu: Add Atom Tremont (Jacobsville) perf/x86/intel/uncore: Add Node ID mask Karsten Graul (7): net/smc: prevent races between smc_lgr_terminate() and smc_conn_free() net/smc: don't wait for send buffer space when data was already sent net/smc: recvmsg and splice_read should return 0 after shutdown net/smc: do not wait under send_lock net/smc: call smc_cdc_msg_send() under send_lock net/smc: use device link provided in qp_context net/smc: fix use of variable in cleared area Kees Cook (2): selftests/seccomp: Enhance per-arch ptrace syscall skip tests exec: load_script: Do not exec truncated interpreter path Keith Busch (3): nvme: lock NS list changes while handling command effects nvme-pci: fix rapid add remove sequence nvme-pci: add missing unlock for reset error Kirill A. Shutemov (1): x86/boot/compressed/64: Do not corrupt EDX on EFER.LME=1 setting Kirill Marinushkin (2): ASoC: pcm3060: Add soft reset on probe ASoC: pcm3060: Add clock select Koen Vandeputte (2): ARM: cns3xxx: Fix writing to wrong PCI config registers after alignment ARM: cns3xxx: Use actual size reads for PCIe Konstantin Khlebnikov (1): inet_diag: fix reporting cgroup classid and fallback to priority Kuninori Morimoto (1): ASoC: simple-card: fixup refcount_t underflow Kyle Tso (1): usb: typec: tcpm: Correct the PPS out_volt calculation Leo (Hanghong) Ma (1): drm/amd/display: Fix MST reboot/poweroff sequence Leonard Crestez (2): PCI: imx: Fix probe failure without power domain PCI: imx: Fix checking pd_pcie_phy device link addition Leonid Iziumtsev (1): dmaengine: imx-dma: fix wrong callback invoke Li RongQing (2): ipv6: propagate genlmsg_reply return code mac80211_hwsim: propagate genlmsg_reply return code Lijun Ou (1): RDMA/hns: Update the kernel header file of hns Linus Torvalds (127): Merge git://git.kernel.org/.../davem/net Merge tag 'gpio-v5.0-3' of git://git.kernel.org/.../linusw/linux-gpio Merge tag 'iommu-fixes-v5.0-rc4' of git://git.kernel.org/.../joro/iommu Merge tag 'sound-5.0-rc5' of git://git.kernel.org/.../tiwai/sound Merge tag 'nfs-for-5.0-3' of git://git.linux-nfs.org/projects/anna/linux-nfs Merge tag 'linux-kselftest-5.0-rc5' of git://git.kernel.org/.../shuah/linux-kselftest Merge tag 'pci-v5.0-fixes-3' of git://git.kernel.org/.../helgaas/pci Merge branch 'linus' of git://git.kernel.org/.../herbert/crypto-2.6 Merge tag 'clk-fixes-for-linus' of git://git.kernel.org/.../clk/linux x86: explicitly align IO accesses in memcpy_{to,from}io Merge tag 'i3c/fixes-for-5.0-rc5' of git://git.kernel.org/.../i3c/linux Merge tag 'mmc-v5.0-rc4' of git://git.kernel.org/.../ulfh/mmc Merge tag 'acpi-5.0-rc5' of git://git.kernel.org/.../rafael/linux-pm Merge tag 'pm-5.0-rc5' of git://git.kernel.org/.../rafael/linux-pm Merge tag 'iomap-5.0-fixes-1' of git://git.kernel.org/pub/scm/fs/xfs/xfs-linux Merge tag 'for-linus' of git://git.kernel.org/.../rdma/rdma Merge tag 'apparmor-pr-2019-02-01' of git://git.kernel.org/.../jj/linux-apparmor Merge tag '5.0-rc4-smb3-fixes' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'arm64-fixes' of git://git.kernel.org/.../arm64/linux Merge tag 'xtensa-20190201' of git://github.com/jcmvbkbc/linux-xtensa Merge branch 'akpm' (patches from Andrew) Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge tag 'for-linus-20190202' of git://git.kernel.dk/linux-block Merge tag 'riscv-for-linus-5.0-rc5' of git://git.kernel.org/.../palmer/riscv-linux Merge tag 'devicetree-fixes-for-5.0-3' of git://git.kernel.org/.../robh/linux Merge tag 'for-5.0-rc4-tag' of git://git.kernel.org/.../kdave/linux Merge branch 'efi-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'smp-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 5.0-rc5 Merge tag 'pinctrl-v5.0-2' of git://git.kernel.org/.../linusw/linux-pinctrl Merge tag 'fuse-fixes-5.0-rc6' of git://git.kernel.org/.../mszeredi/fuse Merge tag 'trace-v5.0-rc3' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'for_linus' of git://git.kernel.org/.../mst/vhost Merge tag 'sound-5.0-rc6' of git://git.kernel.org/.../tiwai/sound Merge branch 'for-linus' of git://git.kernel.org/.../hid/hid Merge tag 'for-5.0/dm-fixes-2' of git://git.kernel.org/.../device-mapper/linux-dm Merge tag 'nfsd-5.0-1' of git://linux-nfs.org/~bfields/linux Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'platform-drivers-x86-v5.0-2' of git://git.infradead.org/linux-platform-drivers-x86 Merge tag 'drm-fixes-2019-02-08' of git://anongit.freedesktop.org/drm/drm Merge tag 'xfs-5.0-fixes-1' of git://git.kernel.org/pub/scm/fs/xfs/xfs-linux Merge tag 'usb-5.0-rc6' of git://git.kernel.org/.../gregkh/usb Merge tag 'tty-5.0-rc6' of git://git.kernel.org/.../gregkh/tty Merge tag 'staging-5.0-rc6' of git://git.kernel.org/.../gregkh/staging Merge tag 'driver-core-5.0-rc6' of git://git.kernel.org/.../gregkh/driver-core Merge tag 'char-misc-5.0-rc6' of git://git.kernel.org/.../gregkh/char-misc Merge git://git.kernel.org/.../davem/net Merge tag 'acpi-5.0-rc6' of git://git.kernel.org/.../rafael/linux-pm Merge tag 'pci-v5.0-fixes-4' of git://git.kernel.org/.../helgaas/pci Merge tag 'iommu-fixes-v5.0-rc5' of git://git.kernel.org/.../joro/iommu Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge branch 'for-linus' of git://git.kernel.org/.../ebiederm/user-namespace Merge tag 'powerpc-5.0-4' of git://git.kernel.org/.../powerpc/linux Merge tag 'arm64-fixes' of git://git.kernel.org/.../arm64/linux Merge tag 'armsoc-fixes-5.0' of git://git.kernel.org/.../soc/soc Merge tag 'for-linus-5.0-rc6-tag' of git://git.kernel.org/.../xen/tip Merge tag 'mtd/fixes-for-5.0-rc6' of git://git.infradead.org/linux-mtd Merge tag 'for-linus-20190209' of git://git.kernel.dk/linux-block Merge tag 'mips_fixes_5.0_3' of git://git.kernel.org/.../mips/linux Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge branch 'locking-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'irq-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge tag 'dmaengine-fix-5.0-rc6' of git://git.infradead.org/users/vkoul/slave-dma Linux 5.0-rc6 Merge branch 'for-linus' of git://git.kernel.org/.../mattst88/alpha Merge tag 's390-5.0-3' of git://git.kernel.org/.../s390/linux Merge branch 'fixes' of git://git.kernel.org/.../evalenti/linux-soc-thermal Merge tag 'ext4_for_linus_stable' of git://git.kernel.org/.../tytso/ext4 Merge tag 'sound-5.0-rc7' of git://git.kernel.org/.../tiwai/sound Merge tag 'riscv-for-linus-5.0-rc7' of git://git.kernel.org/.../palmer/riscv-linux Merge tag 'hwmon-for-v5.0-rc7' of git://git.kernel.org/.../groeck/linux-staging Merge branch 'akpm' (patches from Andrew) Merge tag 'trace-v5.0-rc4' of git://git.kernel.org/.../rostedt/linux-trace Revert "exec: load_script: don't blindly truncate shebang string" Merge branch 'for-linus' of git://git.kernel.org/.../ebiederm/user-namespace Merge git://git.kernel.org/.../davem/net Merge branch 'linus' of git://git.kernel.org/.../herbert/crypto-2.6 Merge tag 'drm-fixes-2019-02-15-1' of git://anongit.freedesktop.org/drm/drm Merge tag 'mmc-v5.0-rc5' of git://git.kernel.org/.../ulfh/mmc Merge tag 'for-5.0/dm-fixes-3' of git://git.kernel.org/.../device-mapper/linux-dm Merge tag 'for-linus-20190215' of git://git.kernel.dk/linux-block Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge tag 'compiler-attributes-for-linus-v5.0-rc7' of git://github.com/ojeda/linux Merge tag 'auxdisplay-for-linus-v5.0-rc7' of git://github.com/ojeda/linux Merge tag 'nfs-for-5.0-4' of git://git.linux-nfs.org/projects/anna/linux-nfs Merge tag 'nfsd-5.0-2' of git://linux-nfs.org/~bfields/linux Merge tag 'armsoc-fixes' of git://git.kernel.org/.../soc/soc Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge tag 'csky-for-linus-5.0-rc6' of git://github.com/c-sky/csky-linux Merge tag 'powerpc-5.0-5' of git://git.kernel.org/.../powerpc/linux Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'efi-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 5.0-rc7 Merge tag 'trace-v5.0-rc4-3' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'for-linus' of git://git.armlinux.org.uk/~rmk/linux-arm Merge tag 'mailbox-fixes-v5.0-rc7' of git://git.linaro.org/landing-teams/working/fujitsu/integration Merge git://git.kernel.org/.../davem/net Merge branch 'fixes-v5.1-rc6' of git://git.kernel.org/.../jmorris/linux-security Merge tag 'mtd/fixes-for-5.0-rc8' of git://git.infradead.org/linux-mtd Merge tag 'gpio-v5.0-4' of git://git.kernel.org/.../linusw/linux-gpio Merge tag 'pinctrl-v5.0-3' of git://git.kernel.org/.../linusw/linux-pinctrl Merge tag 'sound-5.0' of git://git.kernel.org/.../tiwai/sound Merge tag 'docs-5.0-fix' of git://git.lwn.net/linux Merge branch 'akpm' (patches from Andrew) Merge tag 'arm64-fixes' of git://git.kernel.org/.../arm64/linux Merge tag 'ceph-for-5.0-rc8' of git://github.com/ceph/ceph-client Merge tag 'clk-fixes-for-linus' of git://git.kernel.org/.../clk/linux Merge tag 'for-linus' of git://git.kernel.org/.../rdma/rdma Merge tag 'drm-fixes-2019-02-22' of git://anongit.freedesktop.org/drm/drm Merge tag 'kbuild-fixes-v5.0-2' of git://git.kernel.org/.../masahiroy/linux-kbuild Merge branch 'parisc-5.0-1' of git://git.kernel.org/.../deller/parisc-linux Merge tag 'arc-5.0-final' of git://git.kernel.org/.../vgupta/arc Merge tag 'armsoc-fixes' of git://git.kernel.org/.../soc/soc Merge tag 'pm-5.0' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'fixes-v5.0-rc7' of git://git.kernel.org/.../jmorris/linux-security Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge tag 'powerpc-5.0-6' of git://git.kernel.org/.../powerpc/linux Merge git://git.kernel.org/.../davem/net Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Linux 5.0-rc8 Linus Walleij (2): Merge tag 'gpio-5.0-rc4-fixes-for-linus' of git://git.kernel.org/.../brgl/linux into fixes net: phy: realtek: Dummy IRQ calls for RTL8366RB Liu Bo (3): blk-iolatency: fix IO hang due to negative inflight counter Blk-iolatency: warn on negative inflight IO counter blk-mq: remove duplicated definition of blk_mq_freeze_queue Lorenzo Bianconi (8): net: ip_gre: always reports o_key to userspace net: ip6_gre: always reports o_key to userspace mt76x0: eeprom: fix chan_vs_power map in mt76x0_get_power_info net: ipv4: use a dedicated counter for icmp_v4 redirect packets net: ip6_gre: initialize erspan_ver just for erspan tunnels net: ip_gre: do not report erspan_ver for gre or gretap net: ip6_gre: do not report erspan_ver for ip6gre or ip6gretap net: ip6_gre: fix possible NULL pointer dereference in ip6erspan_set_version Loys Ollivier (1): arm64: dts: meson: Fix mmc cd-gpios polarity Lu Baolu (1): iommu/vt-d: Leave scalable mode default off Lubomir Rintel (5): Revert "Input: olpc_apsp - enable the SP clock" Revert "clk: mmp2: add SP clock" Revert "dt-bindings: marvell,mmp2: Add clock id for the SP clock" irqchip/mmp: Only touch the PJ4 IRQ & FIQ bits on enable/disable gpio: pxa: avoid attempting to set pin direction via pinctrl on MMP2 Luca Coelho (1): iwlwifi: make IWLWIFI depend on CFG80211 Lucas De Marchi (1): drm/i915: always return something on DDI clock selection Lucas Stach (1): gpu: ipu-v3: pre: don't trigger update if buffer address doesn't change Lukas Bulwahn (1): MAINTAINERS: unify reference to xen-devel list Lukas Wunner (3): mmc: bcm2835: Fix DMA channel leak on probe error dmaengine: bcm2835: Fix interrupt race on RT dmaengine: bcm2835: Fix abort of transactions Luwei Kang (1): KVM: x86: Recompute PID.ON when clearing PID.SN Lyude Paul (1): drm/i915: Block fbdev HPD processing during suspend Maciej Kwiecien (1): sctp: don't compare hb_timer expire date before starting it Madalin Bucur (1): soc: fsl: qbman: avoid race in clearing QMan interrupt Magnus Karlsson (3): xsk: add missing smp_rmb() in xsk_mmap i40e: fix potential RX buffer starvation for AF_XDP ixgbe: fix potential RX buffer starvation for AF_XDP Manish Chopra (7): qed: Fix bug in tx promiscuous mode settings qed: Fix LACP pdu drops for VFs qed: Fix VF probe failure while FLR qed: Fix system crash in ll2 xmit qed: Fix stack out of bounds bug qed: Fix EQ full firmware assert. qed*: Advance drivers version to 8.37.0.20 Mao Wenan (2): net: crypto set sk to NULL when af_alg_release. net: set static variable an initial value in atl2_probe() Marc Gonzalez (1): ARM: tango: Improve ARCH_MULTIPLATFORM compatibility Marc Zyngier (7): arm64: KVM: Don't generate UNDEF when LORegion feature is present irqchip/gic-v3-its: Plug allocation race for devices sharing a DevID irqchip/gic-v3-its: Gracefully fail on LPI exhaustion net: dsa: Fix lockdep false positive splat arm/arm64: KVM: Allow a VCPU to fully reset itself arm/arm64: KVM: Don't panic on failure to properly reset system registers arm: KVM: Add missing kvm_stage2_has_pmd() helper Maria Pasechnik (1): net/mlx5e: Fix NULL pointer derefernce in set channels error flow Mark Bloch (2): RDMA/mlx5: Fix check for supported user flags when creating a QP RDMA/mlx5: Fix flow creation on representors Mark Brown (6): Merge branch 'for-5.0' of https://git.kernel.org/.../broonie/sound into asoc-5.1 Merge branch 'for-5.0' of https://git.kernel.org/.../broonie/sound into asoc-5.1 for refcount fix Merge branch 'for-5.0' of https://git.kernel.org/.../broonie/sound into asoc-5.1 Merge branch 'for-5.0' of https://git.kernel.org/.../broonie/sound into asoc-5.1 Merge branch 'asoc-5.0' into asoc-linus Merge branch 'asoc-5.1' into asoc-next Mark Rutland (2): perf/core: Don't WARN() for impossible ring-buffer sizes arm64: fix SSBS sanitization Martin Blumenstingl (5): ARM: dts: meson8b: odroidc1: mark the SD card detection GPIO active-low ARM: dts: meson8b: ec100: mark the SD card detection GPIO active-low ARM: dts: meson8m2: mxiii-plus: mark the SD card detection GPIO active-low mmc: meson-gx: fix interrupt name pinctrl: meson: meson8b: fix the sdxc_a data 1..3 pins Martin KaFai Lau (2): bpf: Fix syscall's stackmap lookup potential deadlock bpf: Fix narrow load on a bpf_sock returned from sk_lookup() Martin Kelly (1): tools: iio: iio_generic_buffer: make num_loops signed Martin Kepplinger (1): mtd: rawnand: gpmi: fix MX28 bus master lockup problem Martin Schwidefsky (1): s390/suspend: fix stack setup in swsusp_arch_suspend Martin Wilck (1): scsi: core: reset host byte in DID_NEXUS_FAILURE case Martin Willi (1): esp: Skip TX bytes accounting when sending from a request socket Martynas Pumputis (2): bpf, selftests: fix handling of sparse CPU allocations netfilter: nf_nat: skip nat clash resolution for same-origin entries Masahiro Yamada (3): arch: unexport asm/shmparam.h for all architectures Rename include/{uapi => }/asm-generic/shmparam.h really sh: fix build error for invisible CONFIG_BUILTIN_DTB_SOURCE Masato Suzuki (1): scsi: sd_zbc: Fix sd_zbc_report_zones() buffer allocation Mathias Thore (1): ucc_geth: Reset BQL queue when stopping device Mathieu Desnoyers (1): ARM: 8834/1: Fix: kprobes: optimized kprobes illegal instruction Mathieu Malaterre (1): ASoC: Use __printf markup to silence compiler Matt Ranostay (1): iio: chemical: atlas-ph-sensor: correct IIO_TEMP values to millicelsius Matteo Croce (1): geneve: change NET_UDP_TUNNEL dependency to select Matthew Wilcox (1): drm: Use array_size() when creating lease Matti Kurkela (1): Input: elantech - enable 3rd button support on Fujitsu CELSIUS H780 Mauro Ciancio (1): Input: elan_i2c - add ACPI ID for touchpad in Lenovo V330-15ISK Max Filippov (7): xtensa: SMP: fix ccount_timer_shutdown xtensa: SMP: fix secondary CPU initialization xtensa: smp_lx200_defconfig: fix vectors clash xtensa: SMP: mark each possible CPU as present drivers/irqchip: xtensa: add warning to irq_retrigger drivers/irqchip: xtensa-mx: fix mask and unmask xtensa: SMP: limit number of possible CPUs by NR_CPUS Maxime Chevallier (1): net: phy: marvell10g: Fix Multi-G advertisement to only advertise 10G Meelis Roos (1): alpha: Fix Eiger NR_IRQS to 128 Mel Gorman (1): mm, page_alloc: fix a division by zero error when boosting watermarks v2 Michael Chan (3): bnxt_en: Disable interrupts when allocating CP rings or NQs. bnxt_en: Fix typo in firmware message timeout logic. bnxt_en: Wait longer for the firmware message response to complete. Michael Ellerman (1): powerpc/64s: Fix possible corruption on big endian due to pgd/pud_present() Michael J. Ruhl (1): IB/hfi1: Remove overly conservative VM_EXEC flag check Michael S. Tsirkin (1): virtio: drop internal struct from UAPI Michal Hocko (6): Revert "mm, memory_hotplug: initialize struct pages for the full memory section" mm, memory_hotplug: is_mem_section_removable do not pass the end of a zone mm, memory_hotplug: __offline_pages fix wrong locking proc, oom: do not report alien mms when setting oom_score_adj mm: handle lru_add_drain_all for UP properly mm, memory_hotplug: fix off-by-one in is_pageblock_removable Michal Kalderon (2): qed: Fix iWARP buffer size provided for syn packet processing. qed: Fix iWARP syn packet mac address validation. Michal Soltys (1): bonding: fix PACKET_ORIGDEV regression Miguel Ojeda (4): auxdisplay: ht16k33: fix potential user-after-free on module unload lib/crc32.c: mark crc32_le_base/__crc32c_le_base aliases as __pure Compiler Attributes: add support for __copy (gcc >= 9) include/linux/module.h: copy __init/__exit attrs to init/cleanup_module Mike Marciniszyn (1): IB/hfi1: Add limit test for RC/UC send via loopback Mike Marshall (1): aio: initialize kiocb private in case any filesystems expect it. Mike Snitzer (1): dm: don't use bio_trim() afterall Mikhail Zaslonko (1): mm, memory_hotplug: test_pages_in_a_zone do not pass the end of zone Miklos Szeredi (3): fuse: handle zero sized retrieve correctly cuse: fix ioctl fuse: decrement NR_WRITEBACK_TEMP on the right page Mikulas Patocka (2): dm: add memory barrier before waitqueue_active dm crypt: don't overallocate the integrity tag space Ming Lu (1): scsi: libfc: free skb when receiving invalid flogi resp Moni Shoua (1): net/mlx5: Take lock with IRQs disabled to avoid deadlock Murali Karicheri (1): net: netcp: Fix ethss driver probe issue Nagadheeraj Rottela (1): crypto: cavium/nitrox - Invoke callback after DMA unmap Naoya Horiguchi (1): mm: hwpoison: use do_send_sig_info() instead of force_sig() Naresh Kamboju (1): selftests: netfilter: fix config fragment CONFIG_NF_TABLES_INET Nate Dailey (1): md/raid1: don't clear bitmap bits on interrupted recovery. Nathan Chancellor (3): drm/amdkfd: Fix if preprocessor statement above kfd_fill_iolink_info_for_cpu ASoC: codecs: jz4725b: Remove unnecessary const qualifier arm64/neon: Disable -Wincompatible-pointer-types when building with Clang Neo Hou (2): gpio: sprd: Fix the incorrect data register gpio: sprd: Fix incorrect irq type setting for the async EIC Nicholas Kazlauskas (3): drm/amd/display: Attach VRR properties for eDP connectors drm/amd/display: Expose connector VRR range via debugfs drm/amd/display: Fix negative cursor pos programming Nicolas Morey-Chaisemartin (1): xprtrdma: Make sure Send CQ is allocated on an existing compvec Nicolas Saenz Julienne (1): fpga: stratix10-soc: fix wrong of_node_put() in init function Nikos Tsironis (1): dm thin: fix bug where bio that overwrites thin block ignores FUA Nir Dotan (1): ip6mr: Fix notifiers call on mroute_clean_tables() Oliver O'Halloran (1): powerpc/papr_scm: Use the correct bind address Olivier Moysan (9): ASoC: stm32: i2s: add power management SoC: stm32: i2s: manage clock power ASoC: stm32: i2s: fix IRQ clearing ASoC: stm32: i2s: fix 16 bit format support ASoC: stm32: i2s: fix stream count management ASoC: stm32: i2s: fix dma configuration ASoC: stm32: i2s: remove useless callback ASoC: stm32: i2s: fix race condition in irq handler ASoC: stm32: i2s: skip useless write in slave mode Or Gerlitz (3): net/mlx5e: Move to use common phys port names for vport representors net/mlx5e: Unblock setting vid 0 for VFs through the uplink rep net/mlx5e: Properly set steering match levels for offloaded TC decap rules Oscar Salvador (2): mm, memory_hotplug: don't bail out in do_migrate_range() prematurely mm,memory_hotplug: fix scan_movable_pages() for gigantic hugepages Pablo Neira Ayuso (3): netfilter: nf_tables: unbind set in rule from commit path netfilter: nft_compat: use-after-free when deleting targets netfilter: nf_tables: fix flush after rule deletion in the same batch Palmer Dabbelt (3): RISC-V: defconfig: Move CONFIG_PCI{,E_XILINX} RISC-V: defconfig: Add CRYPTO_DEV_VIRTIO=y Revert "RISC-V: Make BSS section as the last section in vmlinux.lds.S" Pan Bian (1): autofs: drop dentry reference only when it is never used Paolo Abeni (9): bpftool: fix percpu maps updating vsock: cope with memory allocation failure at socket creation time ipv6: route: enforce RCU protection in rt6_update_exception_stamp_rt() ipv6: route: enforce RCU protection in ip6_route_check_nh_onlink() ipv6: route: purge exception on removal udpv6: add the required annotation to mib type fou6: fix proto error handler argument type udpv6: fix possible user after free in error handler udp: fix possible user after free in error handler Paolo Bonzini (3): KVM: x86: work around leak of uninitialized stack contents (CVE-2019-7222) Merge tag 'kvm-arm-fixes-for-5.0' of git://git.kernel.org/.../kvmarm/kvmarm into kvm-master Merge tag 'kvm-s390-master-5.0' of git://git.kernel.org/.../kvms390/linux into kvm-master Paul Burton (5): MIPS: VDSO: Use same -m%-float cflag as the kernel proper MIPS: VDSO: Include $(ccflags-vdso) in o32,n32 .lds builds MIPS: Use lower case for addresses in nexys4ddr.dts MIPS: eBPF: Always return sign extended 32b values MIPS: eBPF: Remove REG_32BIT_ZERO_EX Paul Cercueil (1): MIPS: DTS: jz4740: Correct interrupt number of DMA core Paul Elder (1): usb: gadget: musb: fix short isoc packets with inventra dma Paul Kocialkowski (5): arm64: dts: allwinner: a64: Fix the video engine compatible clk: sunxi-ng: v3s: Fix TCON reset de-assert bit drm/sun4i: tcon: Prepare and enable TCON channel 0 clock at init i2c: bcm2835: Clear current buffer pointers and counts after a transfer net: phy: xgmiitorgmii: Support generic PHY status read Pavel Shilovsky (4): CIFS: Fix possible oops and memory leaks in async IO CIFS: Fix trace command logging for SMB2 reads and writes CIFS: Do not count -ENODATA as failure for query directory CIFS: Do not consider -ENODATA as stat failure for reads Paweł Chmiel (1): Input: pwm-vibra - stop regulator after disabling pwm, not before Peng Hao (1): ARM: pxa: ssp: unneeded to free devm_ allocated data Peter Seiderer (1): tlv320aic32x4: delay i2c access by 1 ms after hardware reset Peter Shier (1): KVM: nVMX: unconditionally cancel preemption timer in free_nested (CVE-2019-7221) Peter Ujfalusi (2): ARM: dts: am335x-evmsk: Fix PHY mode for ethernet ARM: dts: am335x-evm: Fix PHY mode for ethernet Peter Zijlstra (2): perf/x86/intel: Delay memory deallocation until x86_pmu_dead_cpu() x86/mm/cpa: Fix set_mce_nospec() Petr Machata (2): net: cls_flower: Remove filter from mask before freeing it mlxsw: __mlxsw_sp_port_headroom_set(): Fix a use of local variable Petr Vorel (2): apparmor: Fix warning about unused function apparmor_ipv6_postroute doc: Mention MSG_ZEROCOPY implementation for UDP Philip Yang (1): drm/amdgpu: use spin_lock_irqsave to protect vm_manager.pasid_idr Philipp Zabel (1): drm/imx: ignore plane updates on disabled crtcs Pierre Morel (1): s390: vsie: Use effective CRYCBD.31 to check CRYCBD validity Pierre-Louis Bossart (20): ASoC: core: don't increase component module refcount unconditionally ASoC: Intel: Skylake: set .ignore_module_refcount field in component ASoC: hdac_hdmi: use devm_kzalloc for all structures ASoC: Intel: bxt-match: remove prefix for SOF files ASoC: Intel: byt-match.c: remove prefix for SOF files ASoC: Intel: cht-match: remove prefix for SOF files ASoC: Intel: cnl-match: remove prefix for SOF files ASoC: Intel: glk-match: remove prefix for SOF files ASoC: Intel: hda-match: remove prefix for SOF files ASoC: Intel: hsw-bdw-match: remove prefix for SOF files ASoC: Intel: icl-match: remove prefix for SOF files ASoC: Intel: soc-acpi: bxt-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: byt-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: cht-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: glk-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: hsw-bdw-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: icl-match: remove asoc_plat_name field ASoC: Intel: soc-acpi: cnl-match.c: remove asoc_plat_name field ASoC: soc-acpi: remove asoc_plat_name field ASoC: soc-acpi: remove new_mach_data field Piotr Jaroszynski (1): iomap: get/put the page in iomap_page_create/release() Qian Cai (6): x86_64: increase stack size for KASAN_EXTRA mm/hotplug: invalid PFNs from pfn_to_online_page() efi/arm64: Fix debugfs crash by adding a terminator for ptdump marker Revert "mm: use early_pfn_to_nid in page_ext_init" slub: fix SLAB_CONSISTENCY_CHECKS + KASAN_SW_TAGS slub: fix a crash with SLUB_DEBUG + KASAN_SW_TAGS Quentin Perret (1): tracing: Fix number of entries in trace header Raed Salem (1): net/mlx5e: FPGA, fix Innova IPsec TX offload data path performance Rafael J. Wysocki (4): Merge branch 'pm-cpuidle-fixes' Merge branch 'acpi-misc' gpu: drm: radeon: Set DPM_FLAG_NEVER_SKIP when enabling PM-runtime Merge branch 'pm-cpufreq-fixes' Rahul Verma (1): qed: Change verbosity for coalescing message. Rajneesh Bhardwaj (1): x86/CPU: Add Icelake model number Raju Rangoju (2): cxgb4: Export sge_host_page_size to ulds iw_cxgb4: cq/qp mask depends on bar2 pages in a host page Rakesh Pillai (1): mac80211: Restore vif beacon interval if start ap fails Ralph Campbell (1): numa: change get_mempolicy() to use nr_node_ids instead of MAX_NUMNODES Randy Dunlap (1): mtd: rawnand: fix kernel-doc warnings Ravi Bangoria (1): perf mem/c2c: Fix perf_mem_events to support powerpc Rayagonda Kokatanur (1): mailbox: bcm-flexrm-mailbox: Fix FlexRM ring flush timeout issue René van Dorst (1): gpio: MT7621: use a per instance irq_chip structure Rob Herring (1): dt-bindings: Fix dt_binding_check target for in tree builds Robin Murphy (2): ARM: 8835/1: dma-mapping: Clear DMA ops on teardown mm/debug.c: fix __dump_page() for poisoned pages Rodrigo Siqueira (1): drm/vkms: Fix license inconsistent Roger Quadros (1): gpio: pcf857x: Fix interrupts on multiple instances Roman Li (2): drm/amd/display: Fix fclk idle state drm/amd/display: Raise dispclk value for dce11 Ronnie Sahlberg (2): cifs: fix computation for MAX_SMB2_HDR_SIZE cifs: limit amount of data we request for xattrs to CIFSMaxBufSize Ross Lagerwall (1): Revert "scsi: libfc: Add WARN_ON() when deleting rports" Rundong Ge (1): net: dsa: slave: Don't propagate flag changes on down slave interfaces Russell King (8): ARM: iop32x/n2100: fix PCI IRQ mapping Revert "net: phy: marvell: avoid pause mode on SGMII-to-Copper for 88e151x" MAINTAINERS: add maintainer for SFF/SFP/SFP+ support ARM: OMAP2+: fix lack of timer interrupts on CPU1 after hotplug net: sfp: do not probe SFP module before we're attached net: phylink: avoid resolving link state too early net: dsa: fix unintended change of bridge interface STP state net: marvell: mvneta: fix DMA debug warning Ryder Lee (1): pinctrl: mediatek: fix Kconfig build errors for moore core S.j. Wang (1): ASoC: fsl_esai: fix register setting issue in RIGHT_J mode Saeed Mahameed (2): net/mlx4_en: Force CHECKSUM_NONE for short ethernet frames net/mlx5e: XDP, fix redirect resources availability check Salil Mehta (1): net: hns: Fixes the missing put_device in positive leg for roce reset Sandeep Patil (1): mm: proc: smaps_rollup: fix pss_locked calculation Sandy Huang (1): drm/rockchip: rgb: update SPDX license identifier Scott Mayhew (1): sunrpc: fix 4 more call sites that were using stack memory with a scatterlist Sean Christopherson (1): KVM: nVMX: Restore a preemption timer consistency check Sean Tranchetti (1): af_key: unconditionally clone on broadcast Sean Young (1): selftests: Use lirc.h from kernel tree, not from system Sebastian Andrzej Siewior (1): net: dp83640: expire old TX-skb Sebastian Ott (1): Revert "s390/pci: remove bit_lock usage in interrupt handler" Sergei Trofimovich (1): alpha: fix page fault handling for r16-r18 targets Sergej Sawazki (2): ASoC: wm8741: Add digital mute callback ASoC: wm8741: Set OSR mode in hw_params() Shakeel Butt (1): mm, oom: fix use-after-free in oom_kill_process Shawn Guo (1): ARM: dts: vf610-bk4: fix incorrect #address-cells for dspi3 Shay Agroskin (1): net/mlx5e: Fix wrong private flag usage causing checksum disable Shubhrajyoti Datta (1): i2c: cadence: Fix the hold bit setting Sinan Kaya (3): mfd: Fix unmet dependency warning for MFD_TPS68470 platform/x86: Fix unmet dependency warning for ACPI_CMPC platform/x86: Fix unmet dependency warning for SAMSUNG_Q10 Siva Rebbagondla (1): MAINTAINERS: add entry for redpine wireless driver Souptick Joarder (1): arch/arm/xen: Remove duplicate header Srinivas Kandagatla (1): arm64: dts: add msm8996 compatible to gicv3 Stanislav Fomichev (1): bpf/test_run: fix unkillable BPF_PROG_TEST_RUN Stanislaw Gruszka (1): mt76x0u: fix suspend/resume Stefan Agner (1): Input: snvs_pwrkey - allow selecting driver for i.MX 7D Stefan Haberland (1): s390/dasd: fix using offset into zero size array error Stefan O'Rear (1): riscv: Add pte bit to distinguish swap from invalid Stefano Garzarella (2): vsock/virtio: fix kernel panic after device hot-unplug vsock/virtio: reset connected sockets on device removal Steffen Maier (1): scsi: zfcp: fix sysfs block queue limit output for max_segment_size Stephane Eranian (2): perf core: Fix perf_proc_update_handler() bug perf tools: Handle TOPOLOGY headers with no CPU Stephen Boyd (1): Merge tag 'sunxi-clk-fixes-for-5.0' of https://git.kernel.org/.../sunxi/linux into clk-fixes Steve French (1): cifs: update internal module version number Steve Longerbeam (1): gpu: ipu-v3: Fix CSI offsets for imx53 Stuart Henderson (2): ASoC: wm_adsp: Update cached error state on trigger ASoC: wm_adsp: Add support for multiple compressed buffers Su Yanjun (1): vti4: Fix a ipip packet processing bug in 'IPCOMP' virtual tunnel Sudarsana Reddy Kalluru (3): qed: Assign UFP TC value to vlan priority in UFP mode. qed: Consider TX tcs while deriving the max num_queues for PF. qede: Fix system crash on configuring channels. Sudeep Holla (2): ARM: dts: imx: replace gpio-key,wakeup with wakeup-source property firmware: arm_scmi: provide the mandatory device release callback Suravee Suthikulpanit (1): iommu/amd: Fix IOMMU page flush when detach device from a domain Suzuki K Poulose (1): KVM: arm64: Relax the restriction on using stage2 PUD huge mapping Sven Eckelmann (2): batman-adv: Avoid WARN on net_device without parent in netns batman-adv: Force mac header to start of data on xmit Sylwester Nawrocki (31): ASoC: doc: Fix typos in dpcm.rst ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI ASoC: dmaengine: Improve of_node test in dmaengine_pcm_request_chan_of() ASoC: dmaengine: Extend use of chan_names provided in custom DMA config ASoC: samsung: dmaengine: Allow to specify custom DMA device ASoC: samsung: i2s: Convert to single component with multiple DAIs ASoC: samsung: i2s: Restore support for the secondary PCM ASoC: samsung: i2s: Move clk supplier data to common driver data structure ASoC: samsung: i2s: Add widgets and routes for DPCM support ASoC: samsung: i2s: Move core clk to the driver common data structure ASoC: samsung: i2s: Move opclk data to common driver data structure ASoC: samsung: i2s: Move registers cache to common driver data structure ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition ASoC: samsung: i2s: Comments clean up ASoC: samsung: i2s: Convert to SPDX License Indentifier ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: i2s: Prevent potential NULL platform data dereference ASoC: samsung: odroid: Ensure proper sample rate on pri/sec PCM ASoC: samsung: odroid: Add missing DAPM routes ASoC: samsung: i2s: Fix secondary platform device unregistration ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization ASoC: samsung: odroid: Fix of_node refcount unbalance ASoC: samsung: odroid: Prevent uninitialized variable use Takashi Iwai (6): Merge tag 'asoc-fix-v5.0-rc6' of git://git.kernel.org/.../broonie/sound into for-linus ASoC: fsl: Fix of-node refcount unbalance in fsl_ssi_probe_from_dt() ASoC: simple-card: Fix missing of_node_put() at simple_dai_link_of() ASoC: simple-card: Fix of-node refcount unbalance in DAI-link parser ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of() ASoC: qcom: Fix of-node refcount unbalance in qcom_snd_parse_of() Taniya Das (1): clk: qcom: gcc: Use active only source for CPUSS clocks Tariq Toukan (1): net/mlx5: Fix a compilation warning in events.c Tejas Joglekar (1): usb: dwc3: gadget: Handle 0 xfer length for OUT EP Tetsuo Handa (3): block: pass no-op callback to INIT_WORK(). oom, oom_reaper: do not enqueue same task twice fs: ratelimit __find_get_block_slow() failure message. Thadeu Lima de Souza Cascardo (1): selftests: fib_tests: sleep after changing carrier. again. Theodore Ts'o (1): Revert "ext4: use ext4_write_inode() when fsyncing w/o a journal" Thierry Reding (2): mailbox: Export mbox_flush() ARM: tegra: Restore DT ABI on Tegra124 Chromebooks Thinh Nguyen (1): PCI: Work around Synopsys duplicate Device ID (HAPS USB3, NXP i.MX) Thomas Gleixner (2): Merge tag 'irqchip-5.0-3' of git://git.kernel.org/.../maz/arm-platforms into irq/urgent futex: Handle early deadlock return correctly Thomas Hellstrom (4): drm/vmwgfx: Return error code from vmw_execbuf_copy_fence_user drm/vmwgfx: Fix an uninitialized fence handle value drm/vmwgfx: Fix setting of dma masks drm/vmwgfx: Improve on IOMMU detection Thomas Lendacky (1): x86/microcode/amd: Don't falsely trick the late loading mechanism Thomas Petazzoni (1): ARM: dts: armada-xp: fix Armada XP boards NAND description Tina Zhang (1): drm/modes: Prevent division by zero htotal Tiwei Bie (1): virtio: support VIRTIO_F_ORDER_PLATFORM Tobias Brunner (1): xfrm: Fix inbound traffic via XFRM interfaces across network namespaces Toke Høiland-Jørgensen (1): mac80211: Change default tx_sk_pacing_shift to 7 Tomas Winkler (2): mei: me: add ice lake point device id. samples: mei: use /dev/mei0 instead of /dev/mei Tomi Valkeinen (3): drm/omap: dsi: Fix crash in DSI debug dumps drm/omap: dsi: Fix OF platform depopulate drm/omap: dsi: Hack-fix DSI bus flags Tomonori Sakita (1): net: altera_tse: fix msgdma_tx_completion on non-zero fill_level case Tonghao Zhang (2): net/mlx5e: Update hw flows when encap source mac changed net/mlx5e: Don't overwrite pedit action when multiple pedit used Tony Jones (2): perf script: Fix crash when processing recorded stat data perf script python: Add Python3 support to tests/attr.py Tony Lindgren (10): ARM: dts: omap4-droid4: Fix typo in cpcap IRQ flags ARM: dts: Remove unnecessary idle flags for omap5 uart3 clocksource: timer-ti-dm: Fix pwm dmtimer usage of fck reparenting ARM: dts: omap3-gta04: Fix graph_port warning ARM: OMAP5+: Fix inverted nirq pin interrupts with irq_set_type bus: ti-sysc: Fix timer handling with drop pm_runtime_irq_safe() ARM: dts: Configure clock parent for pwm vibra Merge branch 'pwm-dmtimer-fixes' into omap-for-v5.0/fixes-v2 i2c: omap: Use noirq system sleep pm ops to idle device for suspend Merge branch 'am335x-phy-fixes' into omap-for-v5.0/fixes-v2 Tony Luck (1): x86/MCE: Initialize mce.bank in the case of a fatal error in mce_no_way_out() Toshiaki Makita (8): virtio_net: Don't enable NAPI when interface is down virtio_net: Don't call free_old_xmit_skbs for xdp_frames virtio_net: Fix not restoring real_num_rx_queues virtio_net: Fix out of bounds access of sq virtio_net: Don't process redirected XDP frames when XDP is disabled virtio_net: Use xdp_return_frame to free xdp_frames on destroying vqs virtio_net: Differentiate sk_buff and xdp_frame on freeing virtio_net: Account for tx bytes and packets on sending xdp_frames Trond Myklebust (2): NFS: Fix up return value on fatal errors in nfs_page_async_flush() nfsd: Fix error return values for nfsd4_clone_file_range() Tung Nguyen (2): tipc: improve function tipc_wait_for_cond() tipc: improve function tipc_wait_for_rcvmsg() Tuong Lien (1): tipc: fix link session and re-establish issues Tvrtko Ursulin (1): drm/i915/pmu: Fix enable count array size and bounds checking Ulf Hansson (1): wlcore: sdio: Fixup power on/off sequence Ursula Braun (7): net/smc: fix another sizeof to int comparison net/smc: preallocated memory for rdma work requests net/smc: fix sender_free computation net/smc: delete rkey first before switching to unused net/smc: correct state change for peer closing net/smc: fix byte_order for rx_curs_confirmed net/smc: fix smc_poll in SMC_INIT state Vadim Lomovtsev (8): net: thunderx: correct typo in macro name net: thunderx: replace global nicvf_rx_mode_wq work queue for all VFs to private for each of them. net: thunderx: make CFG_DONE message to run through generic send-ack sequence net: thunderx: add nicvf_send_msg_to_pf result check for set_rx_mode_task net: thunderx: rework xcast message structure to make it fit into 64 bit net: thunderx: add mutex to protect mailbox from concurrent calls for same VF net: thunderx: move link state polling function to VF net: thunderx: remove link change polling code and info from nicpf Vaibhav Jain (1): scsi: cxlflash: Prevent deadlock when adapter probe fails Ville Syrjälä (2): drm/i915: Fix skl srckey mask bits drm/i915: Try to sanitize bogus DPLL state left over by broken SNB BIOSen Vincent Chen (1): RISC-V: Add _TIF_NEED_RESCHED check for kernel thread when CONFIG_PREEMPT=y Vincent Guittot (2): PM-runtime: Fix deadlock with ktime_get() PM-runtime: Fix deadlock when canceling hrtimer Vincent Whitchurch (3): mic: vop: Fix broken virtqueues mic: vop: Fix use-after-free on remove mic: vop: Fix crash on remove Vineet Gupta (3): ARC: uacces: remove lp_start, lp_end from clobber list ARCv2: support manual regfile save on interrupts ARCv2: don't assume core 0x54 has dual issue Vinod Koul (1): Merge branch 'fix/brcm' into fixes Viorel Suman (2): ASoC: fsl_spdif: fix TXCLK_DF mask ASoC: fsl_spdif: fix sysclk_df type Vitaly Kuznetsov (2): x86/kvm/nVMX: read from MSR_IA32_VMX_PROCBASED_CTLS2 only when it is available x86/kvm/mmu: fix switch between root and guest MMUs Vivien Didelot (1): net: ethernet: freescale: set FEC ethtool regs version Vladimir Kondratiev (1): mips: cm: reprime error cause Vladimir Murzin (1): arm64: Relax GIC version check during early boot Vladis Dronov (1): HID: debug: fix the ring buffer implementation Waiman Long (4): fs/dcache: Fix incorrect nr_dentry_unused accounting in shrink_dcache_sb() fs: Don't need to put list_lru into its own cacheline fs/dcache: Track & report number of negative dentries Documentation: Fix grammatical error in sysctl/fs.txt & clarify negative dentry Wei Huang (1): x86/boot/compressed/64: Set EFER.LME=1 in 32-bit trampoline before returning to long mode Wei Yongjun (5): ASoC: mediatek: mt8183: make some functions static ASoC: cros_ec_codec: Make symbol 'cros_ec_dai' static ASoC: cs35l36: Make some symbols static mac80211: mesh: fix missing unlock on error in table_path_del() ASoC: wm8741: Make function 'wm8741_mute' static Wen Yang (2): ASoC: stm32: sai: add missing put_device() SoC: imx-sgtl5000: add missing put_device() Will Deacon (1): arm64: ptdump: Don't iterate kernel page tables using PTRS_PER_PXX Willem de Bruijn (3): bpf: only adjust gso_size on bytestream protocols net: validate untrusted gso packets without csum offload net: avoid false positives in untrusted gso validation Xiaoyao Li (1): kvm: vmx: Fix entry number check for add_atomic_switch_msr() Xin Long (3): sctp: check and update stream->out_curr when allocating stream_out sctp: call gso_reset_checksum when computing checksum in sctp_gso_segment sctp: set stream ext to NULL after freeing it in sctp_stream_outq_migrate Yafang Shao (1): bpf: sock recvbuff must be limited by rmem_max in bpf_setsockopt() Yan, Zheng (1): ceph: avoid repeatedly adding inode to mdsc->snap_flush_list Yang Wei (18): net: i825xx: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: alteon: replace dev_kfree_skb_irq by dev_consume_skb_irq net: amd8111e: replace dev_kfree_skb_irq by dev_consume_skb_irq net: apple: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: ti: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: 8139cp: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: caif: call dev_consume_skb_any when skb xmit done net: b44: replace dev_kfree_skb_xxx by dev_consume_skb_xxx for drop profiles net: defxx: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: tulip: de2104x: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: dscc4: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: smsc: epic100: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: fec_mpc52xx: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: fsl_ucc_hdlc: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: sun: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: tehuti: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: via-velocity: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles net: broadcom: replace dev_kfree_skb_irq by dev_consume_skb_irq for drop profiles Yangtao Li (1): cpufreq: scmi: Fix use-after-free in scmi_cpufreq_exit() Yao Liu (1): nfs: Fix NULL pointer dereference of dev_name Yifeng Li (1): mips: loongson64: remove unreachable(), fix loongson_poweroff(). Yishai Hadas (3): IB/uverbs: Fix OOPs upon device disassociation IB/uverbs: Fix ioctl query port to consider device disassociation IB/uverbs: Fix OOPs in uverbs_user_mmap_disassociate Yizhuo (1): ARM: OMAP2+: Variable "reg" in function omap4_dsi_mux_pads() could be uninitialized Yohei Kanemaru (1): ipv6: sr: clear IP6CB(skb) on SRH ip4ip6 encapsulation Yong Zhi (3): ASoC: Intel: Headset button support in broxton machine driver ASoC: Intel: Add Geminilake Dialog Maxim machine driver ASoC: Intel: glk: Add DAI links for Multi-Playback Yonghong Song (2): bpf: btf: allow typedef func_proto tools/bpf: fix test_btf for typedef func_proto case Yonglong Liu (3): net: hns: Fix for missing of_node_put() after of_parse_phandle() net: hns: Restart autoneg need return failed when autoneg off net: hns: Fix wrong read accesses via Clause 45 MDIO protocol Yu Zhang (2): kvm: x86: Return LA57 feature based on hardware capability KVM: MMU: record maximum physical address width in kvm_mmu_extended_role Yu Zhao (1): mm/gup: fix gup_pmd_range() for dax YueHaibing (3): ASoC: cs35l36: Remove unused including <linux/version.h> ASoC: stm32: sai: remove set but not used variables 'mask, cr1' mdio_bus: Fix use-after-free on device_register fails Yufen Yu (1): floppy: check_events callback should not return a negative number Zachary Hays (1): mmc: block: handle complete_work on separate workqueue Zenghui Yu (1): irqchip/gic-v3-its: Fix ITT_entry_size accessor ZhangXiaoxu (1): ipvs: Fix signed integer overflow when setsockopt timeout Zhenzhong Duan (1): x86/speculation: Remove redundant arch_smt_update() invocation Zhiqiang Liu (1): net: fix IPv6 prefix route residue Zhou Yanjie (1): DTS: CI20: Fix bugs in ci20's device tree. shaoyunl (1): drm/amdgpu: Update sdma golden setting for vega20 .clang-format | 43 +- CREDITS | 20 +- Documentation/ABI/stable/sysfs-driver-mlxreg-io | 6 +- Documentation/ABI/testing/sysfs-block | 9 + Documentation/ABI/testing/sysfs-block-zram | 11 +- 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include/sound/soc.h | 73 +- include/trace/events/afs.h | 2 + .../linux/android/{binder_ctl.h => binderfs.h} | 10 +- include/uapi/linux/audit.h | 2 + include/uapi/linux/blkzoned.h | 1 + include/uapi/linux/in.h | 2 +- include/uapi/linux/inet_diag.h | 16 +- include/uapi/linux/input.h | 6 +- include/uapi/linux/ptp_clock.h | 2 +- include/uapi/linux/virtio_config.h | 6 + include/uapi/linux/virtio_ring.h | 10 - include/uapi/rdma/hns-abi.h | 5 + include/uapi/rdma/vmw_pvrdma-abi.h | 1 + include/xen/arm/page-coherent.h | 97 +- init/Kconfig | 14 +- init/initramfs.c | 6 +- init/main.c | 3 +- kernel/bpf/btf.c | 17 +- kernel/bpf/cgroup.c | 3 +- kernel/bpf/hashtab.c | 4 +- kernel/bpf/lpm_trie.c | 1 + kernel/bpf/map_in_map.c | 17 +- kernel/bpf/percpu_freelist.c | 41 +- kernel/bpf/percpu_freelist.h | 4 + kernel/bpf/stackmap.c | 20 +- kernel/bpf/syscall.c | 12 +- kernel/bpf/verifier.c | 72 +- kernel/cpu.c | 38 +- kernel/dma/swiotlb.c | 2 + kernel/events/core.c | 30 +- kernel/events/ring_buffer.c | 3 + kernel/exit.c | 15 +- kernel/fork.c | 14 +- kernel/futex.c | 45 +- kernel/irq/irqdesc.c | 2 +- kernel/irq/manage.c | 3 + kernel/locking/rtmutex.c | 37 +- kernel/locking/rwsem-xadd.c | 11 +- kernel/relay.c | 4 +- kernel/sched/core.c | 19 +- kernel/sched/fair.c | 1 + kernel/sched/psi.c | 23 +- kernel/seccomp.c | 4 + kernel/signal.c | 66 +- kernel/smp.c | 2 - kernel/sys.c | 3 +- kernel/time/posix-cpu-timers.c | 1 + kernel/trace/bpf_trace.c | 14 +- kernel/trace/trace.c | 2 + kernel/trace/trace_kprobe.c | 22 +- kernel/trace/trace_probe_tmpl.h | 6 +- kernel/trace/trace_uprobe.c | 9 +- kernel/umh.c | 33 +- kernel/workqueue.c | 23 + kernel/workqueue_internal.h | 6 +- lib/assoc_array.c | 8 +- lib/crc32.c | 4 +- lib/int_sqrt.c | 2 +- lib/sbitmap.c | 13 +- lib/test_kmod.c | 2 +- lib/test_rhashtable.c | 23 +- lib/test_xarray.c | 57 +- lib/xarray.c | 92 +- mm/backing-dev.c | 1 + mm/debug.c | 4 +- mm/gup.c | 3 +- mm/hugetlb.c | 84 +- mm/kasan/Makefile | 3 + mm/kasan/common.c | 82 +- mm/kasan/tags.c | 2 +- mm/kmemleak.c | 10 +- mm/memblock.c | 11 +- mm/memory-failure.c | 19 +- mm/memory.c | 26 +- mm/memory_hotplug.c | 85 +- mm/mempolicy.c | 6 +- mm/migrate.c | 25 +- mm/mincore.c | 94 +- mm/oom_kill.c | 12 +- mm/page_alloc.c | 40 +- mm/page_ext.c | 4 +- mm/rmap.c | 8 +- mm/shmem.c | 10 +- mm/slab.c | 21 +- mm/slab.h | 7 +- mm/slab_common.c | 3 +- mm/slub.c | 61 +- mm/swap.c | 17 +- mm/usercopy.c | 9 +- mm/userfaultfd.c | 11 +- mm/util.c | 4 +- mm/vmscan.c | 10 - net/ax25/ax25_ip.c | 4 +- net/ax25/ax25_route.c | 19 +- net/batman-adv/bat_v_elp.c | 3 + net/batman-adv/hard-interface.c | 5 +- net/batman-adv/soft-interface.c | 4 + net/bpf/test_run.c | 45 +- net/bpfilter/bpfilter_kern.c | 76 +- net/bpfilter/bpfilter_umh_blob.S | 2 +- net/bridge/br_fdb.c | 5 + net/bridge/br_forward.c | 10 +- net/bridge/br_multicast.c | 9 +- net/bridge/br_netfilter_hooks.c | 2 +- net/bridge/br_netfilter_ipv6.c | 1 + net/bridge/br_private.h | 1 + net/bridge/br_vlan.c | 26 +- net/bridge/netfilter/ebtables.c | 15 +- net/bridge/netfilter/nft_reject_bridge.c | 1 + net/can/bcm.c | 27 + net/can/gw.c | 30 +- net/ceph/ceph_common.c | 11 +- net/ceph/debugfs.c | 2 +- net/ceph/messenger.c | 20 +- net/ceph/osd_client.c | 4 +- net/compat.c | 6 +- net/core/dev.c | 7 +- net/core/filter.c | 48 +- net/core/lwt_bpf.c | 1 + net/core/neighbour.c | 15 +- net/core/skbuff.c | 11 +- net/core/skmsg.c | 3 +- net/core/sock.c | 2 +- net/dccp/ccid.h | 4 +- net/decnet/dn_dev.c | 2 +- net/dsa/master.c | 4 + net/dsa/port.c | 7 +- net/dsa/slave.c | 17 +- net/ipv4/bpfilter/sockopt.c | 58 +- net/ipv4/devinet.c | 2 +- net/ipv4/esp4.c | 2 +- net/ipv4/fib_frontend.c | 4 +- net/ipv4/fib_trie.c | 15 +- net/ipv4/fou.c | 12 +- net/ipv4/gre_demux.c | 17 + net/ipv4/inet_diag.c | 10 +- net/ipv4/inetpeer.c | 1 + net/ipv4/ip_gre.c | 66 +- net/ipv4/ip_input.c | 1 + net/ipv4/ip_sockglue.c | 12 +- net/ipv4/ip_tunnel.c | 8 +- net/ipv4/ip_vti.c | 50 + net/ipv4/netfilter/ipt_CLUSTERIP.c | 2 +- net/ipv4/netfilter/nf_nat_l3proto_ipv4.c | 1 + net/ipv4/netfilter/nf_nat_snmp_basic_main.c | 7 +- net/ipv4/route.c | 7 +- net/ipv4/tcp.c | 4 +- net/ipv4/tcp_ipv4.c | 5 +- net/ipv4/tcp_output.c | 1 + net/ipv4/tcp_timer.c | 2 +- net/ipv4/udp.c | 24 +- net/ipv4/udp_impl.h | 1 + net/ipv4/udplite.c | 1 + net/ipv6/addrconf.c | 11 +- net/ipv6/af_inet6.c | 14 +- net/ipv6/datagram.c | 11 +- net/ipv6/esp6.c | 2 +- net/ipv6/fou6.c | 19 +- net/ipv6/icmp.c | 8 +- net/ipv6/ip6_gre.c | 92 +- net/ipv6/ip6mr.c | 7 +- net/ipv6/netfilter.c | 4 +- net/ipv6/netfilter/nf_nat_l3proto_ipv6.c | 1 + net/ipv6/route.c | 46 +- net/ipv6/seg6.c | 4 +- net/ipv6/seg6_iptunnel.c | 2 + net/ipv6/sit.c | 3 +- net/ipv6/udp.c | 38 +- net/ipv6/udp_impl.h | 1 + net/ipv6/udplite.c | 1 + net/ipv6/xfrm6_tunnel.c | 2 +- net/key/af_key.c | 42 +- net/l2tp/l2tp_core.c | 9 +- net/l2tp/l2tp_core.h | 20 + net/l2tp/l2tp_ip.c | 3 + net/l2tp/l2tp_ip6.c | 3 + net/mac80211/agg-tx.c | 4 +- net/mac80211/cfg.c | 10 +- net/mac80211/main.c | 4 +- net/mac80211/mesh.h | 6 + net/mac80211/mesh_pathtbl.c | 157 +- net/mac80211/rx.c | 13 +- net/mac80211/tx.c | 12 +- net/mac80211/util.c | 6 +- net/netfilter/ipvs/Kconfig | 1 + net/netfilter/ipvs/ip_vs_core.c | 10 +- net/netfilter/ipvs/ip_vs_ctl.c | 23 + net/netfilter/nf_conntrack_core.c | 16 + net/netfilter/nf_flow_table_core.c | 5 +- net/netfilter/nf_tables_api.c | 102 +- net/netfilter/nfnetlink_osf.c | 4 + net/netfilter/nft_compat.c | 180 +- net/netfilter/nft_dynset.c | 18 +- net/netfilter/nft_flow_offload.c | 13 +- net/netfilter/nft_immediate.c | 6 +- net/netfilter/nft_lookup.c | 18 +- net/netfilter/nft_objref.c | 18 +- net/netfilter/x_tables.c | 2 +- net/netrom/nr_timer.c | 20 +- net/openvswitch/flow.c | 8 +- net/openvswitch/flow_netlink.c | 2 +- net/packet/af_packet.c | 9 +- net/phonet/pep.c | 32 +- net/rds/bind.c | 6 +- net/rds/ib_send.c | 4 +- net/rds/message.c | 4 +- net/rds/rds.h | 4 - net/rds/send.c | 2 +- net/rose/rose_route.c | 5 + net/rxrpc/af_rxrpc.c | 70 - net/rxrpc/ar-internal.h | 19 +- net/rxrpc/call_object.c | 97 - net/rxrpc/conn_client.c | 5 +- net/rxrpc/recvmsg.c | 3 +- net/rxrpc/sendmsg.c | 24 +- net/sched/act_tunnel_key.c | 19 +- net/sched/cls_api.c | 3 +- net/sched/cls_flower.c | 25 +- net/sched/cls_tcindex.c | 80 +- net/sched/sch_cake.c | 5 +- net/sched/sch_cbs.c | 3 +- net/sched/sch_drr.c | 7 +- net/sched/sch_dsmark.c | 3 +- net/sched/sch_generic.c | 2 +- net/sched/sch_hfsc.c | 9 +- net/sched/sch_htb.c | 3 +- net/sched/sch_prio.c | 3 +- net/sched/sch_qfq.c | 20 +- net/sched/sch_tbf.c | 3 +- net/sctp/diag.c | 1 + net/sctp/ipv6.c | 8 +- net/sctp/offload.c | 1 + net/sctp/protocol.c | 7 +- net/sctp/sm_make_chunk.c | 11 +- net/sctp/socket.c | 4 +- net/sctp/stream.c | 82 +- net/sctp/transport.c | 3 +- net/smc/af_smc.c | 15 +- net/smc/smc.h | 6 +- net/smc/smc_cdc.c | 23 +- net/smc/smc_cdc.h | 53 +- net/smc/smc_clc.c | 2 +- net/smc/smc_close.c | 9 +- net/smc/smc_core.c | 6 +- net/smc/smc_core.h | 20 + net/smc/smc_ib.c | 6 +- net/smc/smc_llc.c | 3 +- net/smc/smc_pnet.c | 2 +- net/smc/smc_tx.c | 64 +- net/smc/smc_wr.c | 46 +- net/smc/smc_wr.h | 1 + net/socket.c | 82 +- net/sunrpc/auth.c | 3 + net/sunrpc/auth_gss/auth_gss.c | 12 +- net/sunrpc/auth_gss/gss_krb5_seqnum.c | 49 +- net/sunrpc/clnt.c | 20 +- net/sunrpc/debugfs.c | 2 +- net/sunrpc/xprt.c | 3 +- net/sunrpc/xprtrdma/svc_rdma_sendto.c | 105 +- net/sunrpc/xprtrdma/svc_rdma_transport.c | 9 +- net/sunrpc/xprtrdma/verbs.c | 13 +- net/sunrpc/xprtsock.c | 22 + net/tipc/link.c | 17 +- net/tipc/msg.h | 22 + net/tipc/netlink_compat.c | 54 +- net/tipc/node.c | 11 +- net/tipc/socket.c | 11 +- net/tipc/topsrv.c | 2 +- net/tls/tls_sw.c | 6 +- net/unix/af_unix.c | 57 +- net/unix/diag.c | 3 +- net/vmw_vsock/virtio_transport.c | 29 +- net/vmw_vsock/vmci_transport.c | 4 + net/wireless/ap.c | 2 + net/wireless/core.c | 2 + net/wireless/core.h | 2 + net/wireless/nl80211.c | 4 +- net/wireless/pmsr.c | 26 +- net/wireless/reg.c | 13 +- net/wireless/sme.c | 2 +- net/wireless/util.c | 35 +- net/x25/af_x25.c | 19 +- net/xdp/xdp_umem.c | 27 +- net/xdp/xsk.c | 20 +- net/xfrm/xfrm_interface.c | 4 +- net/xfrm/xfrm_policy.c | 67 +- net/xfrm/xfrm_state.c | 30 +- net/xfrm/xfrm_user.c | 15 +- samples/bpf/Makefile | 1 + samples/bpf/asm_goto_workaround.h | 16 + samples/bpf/test_cgrp2_attach2.c | 14 +- samples/bpf/test_current_task_under_cgroup_user.c | 2 +- samples/bpf/xdp1_user.c | 2 +- samples/mei/mei-amt-version.c | 2 +- samples/seccomp/Makefile | 1 + scripts/Kbuild.include | 4 - scripts/coccinelle/api/alloc/alloc_cast.cocci | 8 +- scripts/coccinelle/api/alloc/zalloc-simple.cocci | 11 +- scripts/gcc-plugins/arm_ssp_per_task_plugin.c | 23 +- scripts/kallsyms.c | 4 +- scripts/kconfig/Makefile | 2 +- scripts/mod/modpost.c | 2 +- security/apparmor/domain.c | 5 +- security/apparmor/lsm.c | 2 + security/keys/internal.h | 13 +- security/keys/key.c | 5 +- security/keys/keyctl.c | 1 + security/keys/keyring.c | 4 +- security/keys/proc.c | 3 +- security/keys/process_keys.c | 1 + security/keys/request_key.c | 73 +- security/keys/request_key_auth.c | 18 +- security/lsm_audit.c | 10 +- security/security.c | 7 + security/selinux/ss/policydb.c | 3 +- security/yama/yama_lsm.c | 4 +- sound/aoa/soundbus/i2sbus/core.c | 4 +- sound/pci/hda/patch_realtek.c | 42 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 67 +- sound/soc/codecs/Makefile | 14 + sound/soc/codecs/ad193x.c | 76 +- sound/soc/codecs/ad193x.h | 8 + sound/soc/codecs/adau1977.c | 12 +- sound/soc/codecs/adau7002.c | 45 + sound/soc/codecs/ak4118.c | 1 + sound/soc/codecs/ak4458.c | 79 +- sound/soc/codecs/cros_ec_codec.c | 441 ++ sound/soc/codecs/cs35l36.c | 1957 ++++++++ sound/soc/codecs/cs35l36.h | 446 ++ sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs4341.c | 346 ++ sound/soc/codecs/cs47l24.c | 52 +- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/da7219.c | 197 +- sound/soc/codecs/da7219.h | 2 + sound/soc/codecs/dmic.c | 4 +- sound/soc/codecs/es8316.c | 200 +- sound/soc/codecs/es8316.h | 7 + sound/soc/codecs/hdac_hdmi.c | 87 +- sound/soc/codecs/jz4725b.c | 598 +++ sound/soc/codecs/jz4740.c | 26 +- sound/soc/codecs/max98090.c | 15 - sound/soc/codecs/max98373.c | 6 +- sound/soc/codecs/max9860.c | 3 +- sound/soc/codecs/max98927.c | 8 +- sound/soc/codecs/msm8916-wcd-analog.c | 3 + sound/soc/codecs/msm8916-wcd-digital.c | 23 +- sound/soc/codecs/mt6351.c | 4 +- sound/soc/codecs/mt6358.c | 2336 +++++++++ sound/soc/codecs/mt6358.h | 2314 +++++++++ sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 1 + sound/soc/codecs/pcm186x.c | 8 +- sound/soc/codecs/pcm3060.c | 35 + sound/soc/codecs/pcm3060.h | 5 + sound/soc/codecs/pcm512x.c | 37 +- sound/soc/codecs/rk3328_codec.c | 519 ++ sound/soc/codecs/rk3328_codec.h | 210 + sound/soc/codecs/rl6347a.c | 4 +- sound/soc/codecs/rt274.c | 19 +- sound/soc/codecs/rt286.c | 8 +- sound/soc/codecs/rt298.c | 12 +- sound/soc/codecs/rt5514.c | 1 + sound/soc/codecs/rt5640.c | 6 +- sound/soc/codecs/rt5645.c | 66 +- sound/soc/codecs/rt5651.c | 67 +- sound/soc/codecs/rt5651.h | 1 + sound/soc/codecs/rt5670.c | 54 - sound/soc/codecs/rt5677.c | 8 +- sound/soc/codecs/rt5682.c | 8 + sound/soc/codecs/sgtl5000.c | 19 +- sound/soc/codecs/ssm2602.c | 71 +- sound/soc/codecs/tlv320aic31xx.c | 16 +- sound/soc/codecs/tlv320aic32x4.c | 31 +- sound/soc/codecs/tlv320aic32x4.h | 1 + sound/soc/codecs/tlv320aic3x.c | 25 +- sound/soc/codecs/tscs42xx.c | 2 +- sound/soc/codecs/wcd-clsh-v2.c | 576 +++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5244 ++++++++++++++++++++ sound/soc/codecs/wcd9335.h | 640 +++ sound/soc/codecs/wm5102.c | 64 +- sound/soc/codecs/wm5110.c | 80 +- sound/soc/codecs/wm8741.c | 22 +- sound/soc/codecs/wm8770.c | 18 +- sound/soc/codecs/wm8904.c | 21 +- sound/soc/codecs/wm8962.c | 9 +- sound/soc/codecs/wm8995.c | 29 +- sound/soc/codecs/wm8996.c | 9 +- sound/soc/codecs/wm8997.c | 60 +- sound/soc/codecs/wm8998.c | 60 +- sound/soc/codecs/wm_adsp.c | 498 +- sound/soc/codecs/wm_adsp.h | 4 +- sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl-asoc-card.c | 1 + sound/soc/fsl/fsl_esai.c | 7 +- sound/soc/fsl/fsl_micfil.c | 826 +++ sound/soc/fsl/fsl_micfil.h | 283 ++ sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/fsl_spdif.c | 12 +- sound/soc/fsl/fsl_spdif.h | 2 +- sound/soc/fsl/fsl_ssi.c | 6 +- sound/soc/fsl/imx-sgtl5000.c | 12 +- sound/soc/fsl/imx-spdif.c | 4 +- sound/soc/generic/Kconfig | 17 - sound/soc/generic/Makefile | 4 - sound/soc/generic/audio-graph-card.c | 481 +- sound/soc/generic/audio-graph-scu-card.c | 501 -- sound/soc/generic/simple-card-utils.c | 25 +- sound/soc/generic/simple-card.c | 479 +- sound/soc/generic/simple-scu-card.c | 474 -- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 45 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 3 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 +- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bdw-rt5677.c | 14 + sound/soc/intel/boards/broadwell.c | 16 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 114 +- sound/soc/intel/boards/bxt_rt298.c | 13 + sound/soc/intel/boards/bytcht_da7213.c | 8 + sound/soc/intel/boards/bytcht_es8316.c | 336 +- sound/soc/intel/boards/bytcr_rt5640.c | 23 +- sound/soc/intel/boards/bytcr_rt5651.c | 180 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 + sound/soc/intel/boards/cht_bsw_nau8824.c | 12 + sound/soc/intel/boards/cht_bsw_rt5645.c | 9 + sound/soc/intel/boards/cht_bsw_rt5672.c | 30 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 23 +- sound/soc/intel/boards/haswell.c | 15 + sound/soc/intel/boards/kbl_da7219_max98927.c | 207 +- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 25 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 92 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 70 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 5 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 10 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 2 +- .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 20 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 5 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 19 + sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 4 +- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/mediatek/Kconfig | 19 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/Makefile | 2 + sound/soc/mediatek/common/mtk-btcvsd.c | 1364 +++++ sound/soc/mediatek/mt8183/Makefile | 13 + sound/soc/mediatek/mt8183/mt8183-afe-clk.c | 611 +++ sound/soc/mediatek/mt8183/mt8183-afe-clk.h | 38 + sound/soc/mediatek/mt8183/mt8183-afe-common.h | 108 + sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 1237 +++++ sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 501 ++ sound/soc/mediatek/mt8183/mt8183-dai-hostless.c | 118 + sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 1040 ++++ sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 318 ++ sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 639 +++ sound/soc/mediatek/mt8183/mt8183-interconnection.h | 33 + sound/soc/mediatek/mt8183/mt8183-reg.h | 1666 +++++++ sound/soc/pxa/Makefile | 1 - sound/soc/qcom/Kconfig | 4 +- sound/soc/qcom/apq8016_sbc.c | 21 +- sound/soc/qcom/apq8096.c | 71 +- sound/soc/qcom/common.c | 10 +- sound/soc/qcom/sdm845.c | 2 + sound/soc/samsung/dma.h | 3 +- sound/soc/samsung/dmaengine.c | 16 +- sound/soc/samsung/i2s.c | 725 +-- sound/soc/samsung/odroid.c | 206 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 40 +- sound/soc/sh/rcar/gen.c | 24 + sound/soc/sh/rcar/rsnd.h | 27 + sound/soc/sh/rcar/src.c | 125 +- sound/soc/sh/rcar/ssiu.c | 24 +- sound/soc/soc-compress.c | 177 +- sound/soc/soc-core.c | 262 +- sound/soc/soc-dapm.c | 271 +- sound/soc/soc-generic-dmaengine-pcm.c | 13 +- sound/soc/soc-pcm.c | 4 + sound/soc/soc-topology.c | 174 +- sound/soc/sprd/Kconfig | 6 + sound/soc/sprd/Makefile | 4 + sound/soc/sprd/sprd-pcm-dma.c | 562 +++ sound/soc/sprd/sprd-pcm-dma.h | 15 + sound/soc/stm/stm32_i2s.c | 136 +- sound/soc/stm/stm32_sai.c | 11 +- sound/soc/stm/stm32_sai_sub.c | 8 +- sound/soc/sunxi/sun4i-codec.c | 149 +- sound/soc/sunxi/sun50i-codec-analog.c | 4 +- sound/soc/ti/davinci-mcasp.c | 175 +- sound/soc/xilinx/Kconfig | 14 + sound/soc/xilinx/Makefile | 4 + sound/soc/xilinx/xlnx_formatter_pcm.c | 708 +++ sound/soc/xilinx/xlnx_spdif.c | 339 ++ sound/sparc/dbri.c | 4 +- tools/arch/powerpc/include/uapi/asm/perf_regs.h | 1 + tools/arch/powerpc/include/uapi/asm/unistd.h | 404 -- tools/arch/riscv/include/uapi/asm/bitsperlong.h | 25 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virt/kvm/arm/vgic/vgic-mmio.c | 34 +- virt/kvm/arm/vgic/vgic-v2.c | 4 +- virt/kvm/arm/vgic/vgic-v3.c | 8 +- virt/kvm/arm/vgic/vgic.c | 118 +- virt/kvm/kvm_main.c | 12 +- 2137 files changed, 52369 insertions(+), 14309 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt create mode 100644 Documentation/devicetree/bindings/sound/cs35l36.txt create mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4725b-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/mt6358.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8183-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mtk-btcvsd-snd.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt delete mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt create mode 100644 Documentation/devicetree/bindings/sound/sprd-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,audio-formatter.txt create mode 100644 Documentation/devicetree/bindings/sound/xlnx,spdif.txt create mode 100644 arch/arm/mach-tango/pm.h create mode 100644 drivers/reset/reset-socfpga.c rename drivers/reset/{reset-uniphier-usb3.c => reset-uniphier-glue.c} (73%) create mode 100644 drivers/tty/serial/earlycon-riscv-sbi.c delete mode 100644 drivers/usb/usbip/README rename include/{uapi => }/asm-generic/shmparam.h (100%) create mode 100644 include/keys/request_key_auth-type.h create mode 100644 include/sound/cs35l36.h rename include/uapi/linux/android/{binder_ctl.h => binderfs.h} (83%) create mode 100644 samples/bpf/asm_goto_workaround.h create mode 100644 sound/soc/codecs/cros_ec_codec.c create mode 100644 sound/soc/codecs/cs35l36.c create mode 100644 sound/soc/codecs/cs35l36.h create mode 100644 sound/soc/codecs/cs4341.c create mode 100644 sound/soc/codecs/jz4725b.c create mode 100644 sound/soc/codecs/mt6358.c create mode 100644 sound/soc/codecs/mt6358.h create mode 100644 sound/soc/codecs/rk3328_codec.c create mode 100644 sound/soc/codecs/rk3328_codec.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c create mode 100644 sound/soc/codecs/wcd9335.h create mode 100644 sound/soc/fsl/fsl_micfil.c create mode 100644 sound/soc/fsl/fsl_micfil.h delete mode 100644 sound/soc/generic/audio-graph-scu-card.c delete mode 100644 sound/soc/generic/simple-scu-card.c create mode 100644 sound/soc/mediatek/common/mtk-btcvsd.c create mode 100644 sound/soc/mediatek/mt8183/Makefile create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-clk.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-clk.h create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-common.h create mode 100644 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-adda.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8183/mt8183-interconnection.h create mode 100644 sound/soc/mediatek/mt8183/mt8183-reg.h create mode 100644 sound/soc/sprd/Kconfig create mode 100644 sound/soc/sprd/Makefile create mode 100644 sound/soc/sprd/sprd-pcm-dma.c create mode 100644 sound/soc/sprd/sprd-pcm-dma.h create mode 100644 sound/soc/xilinx/xlnx_formatter_pcm.c create mode 100644 sound/soc/xilinx/xlnx_spdif.c delete mode 100644 tools/arch/powerpc/include/uapi/asm/unistd.h create mode 100644 tools/arch/riscv/include/uapi/asm/bitsperlong.h create mode 100644 tools/include/uapi/linux/mount.h create mode 100644 tools/include/uapi/linux/pkt_sched.h create mode 100644 tools/perf/arch/arm/tests/vectors-page.c create mode 100644 tools/perf/arch/powerpc/entry/syscalls/syscall.tbl create mode 100644 tools/perf/arch/powerpc/util/mem-events.c rename tools/perf/util/{find-vdso-map.c => find-map.c} (71%) create mode 100644 tools/testing/selftests/filesystems/binderfs/.gitignore create mode 100644 tools/testing/selftests/filesystems/binderfs/Makefile create mode 100644 tools/testing/selftests/filesystems/binderfs/binderfs_test.c create mode 100644 tools/testing/selftests/filesystems/binderfs/config create mode 100755 tools/testing/selftests/netfilter/nft_nat.sh create mode 100644 tools/testing/selftests/proc/setns-dcache.c

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Set sample_type in base config
by Jenny TC 28 Feb '19

28 Feb '19

From: Jenny TC <jenny.tc(a)intel.com> sample_type defined in topology 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 is reflected int eh 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 to 1 helps to fix this. 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 b0e6fb9..9260e9a 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

3 2

[alsa-devel] [RFC PATCH 1/2] ASoC: simple-card: add support for bclk_ratio
by Sven Van Asbroeck 27 Feb '19

27 Feb '19

In some situations, codec configuration will depend on the bclk_ratio generated by the cpu dai. The tda998x hdmi transmitter is an example of this. Allow simple-card to set the bclk_ratio via the 'bclk-slot-ratio' devicetree property, which describes the bclk to slot rate ratio. This value is converted to the bclk to sample ratio before being passed into set_bclk_ratio(). Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- sound/soc/generic/simple-card.c | 33 ++++++++++++++++++++++++++++++++- 1 file changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3fe34417ec89..61e9ba4e9b58 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -25,6 +25,7 @@ struct simple_card_data { struct asoc_simple_card_data adata; struct snd_soc_codec_conf *codec_conf; unsigned int mclk_fs; + unsigned int bclk_slot_ratio; } *dai_props; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; @@ -97,7 +98,7 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct simple_card_data *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; + unsigned int mclk, bclk_ratio, mclk_fs = 0, bclk_slot_ratio = 0; int ret = 0; if (dai_props->mclk_fs) @@ -124,6 +125,23 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, if (ret && ret != -ENOTSUPP) goto err; } + + if (dai_props->bclk_slot_ratio) + bclk_slot_ratio = dai_props->bclk_slot_ratio; + + if (bclk_slot_ratio) { + /* FIXME do we need to care about TDM slots here ? */ + bclk_ratio = snd_soc_calc_frame_size(bclk_slot_ratio, + params_channels(params), 1); + + ret = snd_soc_dai_set_bclk_ratio(codec_dai, bclk_ratio); + if (ret && ret != -ENOTSUPP) + goto err; + + ret = snd_soc_dai_set_bclk_ratio(cpu_dai, bclk_ratio); + if (ret && ret != -ENOTSUPP) + goto err; + } return 0; err: return ret; @@ -277,6 +295,12 @@ static int asoc_simple_card_dai_link_of_dpcm(struct device_node *top, of_property_read_u32(node, prop, &dai_props->mclk_fs); of_property_read_u32(np, prop, &dai_props->mclk_fs); + snprintf(prop, sizeof(prop), "%sbclk-slot-ratio", prefix); + of_property_read_u32(top, PREFIX "bclk-slot-ratio", + &dai_props->bclk_slot_ratio); + of_property_read_u32(node, prop, &dai_props->bclk_slot_ratio); + of_property_read_u32(np, prop, &dai_props->bclk_slot_ratio); + ret = asoc_simple_card_parse_daifmt(dev, node, codec, prefix, &dai_link->dai_fmt); if (ret < 0) @@ -349,6 +373,13 @@ static int asoc_simple_card_dai_link_of(struct device_node *top, of_property_read_u32(cpu, prop, &dai_props->mclk_fs); of_property_read_u32(codec, prop, &dai_props->mclk_fs); + snprintf(prop, sizeof(prop), "%sbclk-slot-ratio", prefix); + of_property_read_u32(top, PREFIX "bclk-slot-ratio", + &dai_props->bclk_slot_ratio); + of_property_read_u32(node, prop, &dai_props->bclk_slot_ratio); + of_property_read_u32(cpu, prop, &dai_props->bclk_slot_ratio); + of_property_read_u32(codec, prop, &dai_props->bclk_slot_ratio); + ret = asoc_simple_card_parse_cpu(cpu, dai_link, DAI, CELL, &single_cpu); if (ret < 0) -- 2.17.1

4 12

Re: [alsa-devel] increase number of alsa loopback
by Jaroslav Kysela 27 Feb '19

27 Feb '19

Dne 27. 02. 19 v 16:44 Olivier SCHMITT napsal(a): > Hello, > > Thanks for reply, > > I have configure /etc/modprobe.d/alsa-base.conf with: > > options snd-aloop > index=5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29 > id=lp0,lp1,lp2,lp3,lp4,lp5,lp6,lp7,lp8,lp9,lp10,lp11,lp12,lp13,lp14,lp15,lp16,lp17,lp18,lp19,lp20,lp21,lp22,lp23,lp24 > enable=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 > pcm_substreams=2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2,2 > > but when i use index more than 32, i have some message in syslog that > it's say alsa can't use index greather than 32... > > Maybe have your an idea for modify et recompiling kernel? I'm afraid, the ALSA card limit (32) is hardcoded and cannot be easily changed. The recompilation won't help. You may use up to 8 pcm substreams per the loop card. This limit can be increased by changing the value in aloop.c (kernel driver). Jaroslav > Le 27/02/2019 à 15:35, Jaroslav Kysela a écrit : >> Dne 27. 02. 19 v 15:20 Olivier SCHMITT napsal(a): >>> Hello All, >>> >>> First, thanks for your project! >>> >>> I search to increase the maximum number of 32 alsa loopback. >>> >>> Can someone say me how do this? >> There's hardcoded limit for the PCM substreams to 8, but you may enable >> more loopback cards (options snd-aloop enable=1,1,1,1). >> >> Jaroslav >> -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] increase number of alsa loopback
by Olivier SCHMITT 27 Feb '19

27 Feb '19

Hello All, First, thanks for your project! I search to increase the maximum number of 32 alsa loopback. Can someone say me how do this? Thanks in advance! Olivier.

2 1

[alsa-devel] [PATCH] ASoC: tlv320aix3x: fix reset gpio reference counting
by Philipp Puschmann 27 Feb '19

27 Feb '19

This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> --- sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Set sample_type in base config
by Jenny TC 27 Feb '19

27 Feb '19

sample_type defined in topology 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 is reflected int eh 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 to 1 helps to fix this. 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 b0e6fb9..9260e9a 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

1 0

[alsa-devel] [PATCH V3] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 27 Feb '19

27 Feb '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> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - initialize tx_mask and rx_mask in probe sound/soc/fsl/fsl_esai.c | 35 +++++++++++++++++++++++++---------- 1 file changed, 25 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..7ee91c28b636 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: @@ -611,12 +606,23 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, 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 +912,15 @@ 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 2

[alsa-devel] [asoc:for-5.1 263/274] sound/soc/codecs/wm_adsp.c:3441:23: sparse: warning: restricted __be32 degrades to integer
by kbuild test robot 27 Feb '19

27 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.1 head: 7b6b0049e2b70d103adf1b7d0320802f70ddceca commit: 4f2d4eabf57718875b97363a3bd35de490f354c5 [263/274] ASoC: wm_adsp: Add support for multiple compressed buffers reproduce: # apt-get install sparse git checkout 4f2d4eabf57718875b97363a3bd35de490f354c5 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' All warnings (new ones prefixed by >>): sound/soc/codecs/wm_adsp.c:880:19: sparse: warning: incorrect type in initializer (different base types) sound/soc/codecs/wm_adsp.c:880:19: sparse: expected unsigned int [usertype] val sound/soc/codecs/wm_adsp.c:880:19: sparse: got restricted __be32 [usertype] sound/soc/codecs/wm_adsp.c:1586:22: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/soc/codecs/wm_adsp.c:2057:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2057:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2057:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2078:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2078:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2078:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2171:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2171:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2171:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2192:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2192:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2192:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2213:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2213:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2213:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3242:14: sparse: warning: incorrect type in assignment (different base types) sound/soc/codecs/wm_adsp.c:3242:14: sparse: expected unsigned int [usertype] data sound/soc/codecs/wm_adsp.c:3242:14: sparse: got restricted __be32 [usertype] sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3440:27: sparse: warning: incorrect type in assignment (different base types) sound/soc/codecs/wm_adsp.c:3440:27: sparse: expected restricted __be32 [addressable] [usertype] versions sound/soc/codecs/wm_adsp.c:3440:27: sparse: got unsigned int >> sound/soc/codecs/wm_adsp.c:3441:23: sparse: warning: restricted __be32 degrades to integer sound/soc/codecs/wm_adsp.c:3452:34: sparse: warning: incorrect type in assignment (different base types) sound/soc/codecs/wm_adsp.c:3452:34: sparse: expected restricted __be32 sound/soc/codecs/wm_adsp.c:3452:34: sparse: got unsigned int sparse warnings: (new ones prefixed by >>) sound/soc/codecs/wm_adsp.c:880:19: sparse: warning: incorrect type in initializer (different base types) sound/soc/codecs/wm_adsp.c:880:19: sparse: expected unsigned int [usertype] val sound/soc/codecs/wm_adsp.c:880:19: sparse: got restricted __be32 [usertype] sound/soc/codecs/wm_adsp.c:1586:22: sparse: warning: restricted snd_ctl_elem_type_t degrades to integer sound/soc/codecs/wm_adsp.c:2057:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2057:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2057:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2078:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2078:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2078:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2171:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2171:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2171:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2192:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2192:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2192:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:2213:54: sparse: warning: incorrect type in argument 8 (different base types) sound/soc/codecs/wm_adsp.c:2213:54: sparse: expected unsigned int type sound/soc/codecs/wm_adsp.c:2213:54: sparse: got restricted snd_ctl_elem_type_t [usertype] sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3220:27: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3242:14: sparse: warning: incorrect type in assignment (different base types) sound/soc/codecs/wm_adsp.c:3242:14: sparse: expected unsigned int [usertype] data sound/soc/codecs/wm_adsp.c:3242:14: sparse: got restricted __be32 [usertype] sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3420:29: sparse: warning: cast to restricted __be32 sound/soc/codecs/wm_adsp.c:3440:27: sparse: warning: incorrect type in assignment (different base types) >> sound/soc/codecs/wm_adsp.c:3440:27: sparse: expected restricted __be32 [addressable] [usertype] versions >> sound/soc/codecs/wm_adsp.c:3440:27: sparse: got unsigned int sound/soc/codecs/wm_adsp.c:3441:23: sparse: warning: restricted __be32 degrades to integer sound/soc/codecs/wm_adsp.c:3452:34: sparse: warning: incorrect type in assignment (different base types) >> sound/soc/codecs/wm_adsp.c:3452:34: sparse: expected restricted __be32 sound/soc/codecs/wm_adsp.c:3452:34: sparse: got unsigned int vim +3441 sound/soc/codecs/wm_adsp.c 3230 3231 static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, 3232 unsigned int mem_addr, u32 data) 3233 { 3234 struct wm_adsp_region const *mem = wm_adsp_find_region(dsp, mem_type); 3235 unsigned int reg; 3236 3237 if (!mem) 3238 return -EINVAL; 3239 3240 reg = wm_adsp_region_to_reg(mem, mem_addr); 3241 > 3242 data = cpu_to_be32(data & 0x00ffffffu); 3243 3244 return regmap_raw_write(dsp->regmap, reg, &data, sizeof(data)); 3245 } 3246 3247 static inline int wm_adsp_buffer_read(struct wm_adsp_compr_buf *buf, 3248 unsigned int field_offset, u32 *data) 3249 { 3250 return wm_adsp_read_data_word(buf->dsp, buf->host_buf_mem_type, 3251 buf->host_buf_ptr + field_offset, data); 3252 } 3253 3254 static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, 3255 unsigned int field_offset, u32 data) 3256 { 3257 return wm_adsp_write_data_word(buf->dsp, buf->host_buf_mem_type, 3258 buf->host_buf_ptr + field_offset, data); 3259 } 3260 3261 static void wm_adsp_remove_padding(u32 *buf, int nwords, int data_word_size) 3262 { 3263 u8 *pack_in = (u8 *)buf; 3264 u8 *pack_out = (u8 *)buf; 3265 int i, j; 3266 3267 /* Remove the padding bytes from the data read from the DSP */ 3268 for (i = 0; i < nwords; i++) { 3269 for (j = 0; j < data_word_size; j++) 3270 *pack_out++ = *pack_in++; 3271 3272 pack_in += sizeof(*buf) - data_word_size; 3273 } 3274 } 3275 3276 static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) 3277 { 3278 const struct wm_adsp_fw_caps *caps = wm_adsp_fw[buf->dsp->fw].caps; 3279 struct wm_adsp_buffer_region *region; 3280 u32 offset = 0; 3281 int i, ret; 3282 3283 buf->regions = kcalloc(caps->num_regions, sizeof(*buf->regions), 3284 GFP_KERNEL); 3285 if (!buf->regions) 3286 return -ENOMEM; 3287 3288 for (i = 0; i < caps->num_regions; ++i) { 3289 region = &buf->regions[i]; 3290 3291 region->offset = offset; 3292 region->mem_type = caps->region_defs[i].mem_type; 3293 3294 ret = wm_adsp_buffer_read(buf, caps->region_defs[i].base_offset, 3295 &region->base_addr); 3296 if (ret < 0) 3297 return ret; 3298 3299 ret = wm_adsp_buffer_read(buf, caps->region_defs[i].size_offset, 3300 &offset); 3301 if (ret < 0) 3302 return ret; 3303 3304 region->cumulative_size = offset; 3305 3306 adsp_dbg(buf->dsp, 3307 "region=%d type=%d base=%08x off=%08x size=%08x\n", 3308 i, region->mem_type, region->base_addr, 3309 region->offset, region->cumulative_size); 3310 } 3311 3312 return 0; 3313 } 3314 3315 static void wm_adsp_buffer_clear(struct wm_adsp_compr_buf *buf) 3316 { 3317 buf->irq_count = 0xFFFFFFFF; 3318 buf->read_index = -1; 3319 buf->avail = 0; 3320 } 3321 3322 static struct wm_adsp_compr_buf *wm_adsp_buffer_alloc(struct wm_adsp *dsp) 3323 { 3324 struct wm_adsp_compr_buf *buf; 3325 3326 buf = kzalloc(sizeof(*buf), GFP_KERNEL); 3327 if (!buf) 3328 return NULL; 3329 3330 buf->dsp = dsp; 3331 3332 wm_adsp_buffer_clear(buf); 3333 3334 list_add_tail(&buf->list, &dsp->buffer_list); 3335 3336 return buf; 3337 } 3338 3339 static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) 3340 { 3341 struct wm_adsp_alg_region *alg_region; 3342 struct wm_adsp_compr_buf *buf; 3343 u32 xmalg, addr, magic; 3344 int i, ret; 3345 3346 buf = wm_adsp_buffer_alloc(dsp); 3347 if (!buf) 3348 return -ENOMEM; 3349 3350 alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); 3351 xmalg = sizeof(struct wm_adsp_system_config_xm_hdr) / sizeof(__be32); 3352 3353 addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); 3354 ret = wm_adsp_read_data_word(dsp, WMFW_ADSP2_XM, addr, &magic); 3355 if (ret < 0) 3356 return ret; 3357 3358 if (magic != WM_ADSP_ALG_XM_STRUCT_MAGIC) 3359 return -ENODEV; 3360 3361 addr = alg_region->base + xmalg + ALG_XM_FIELD(host_buf_ptr); 3362 for (i = 0; i < 5; ++i) { 3363 ret = wm_adsp_read_data_word(dsp, WMFW_ADSP2_XM, addr, 3364 &buf->host_buf_ptr); 3365 if (ret < 0) 3366 return ret; 3367 3368 if (buf->host_buf_ptr) 3369 break; 3370 3371 usleep_range(1000, 2000); 3372 } 3373 3374 if (!buf->host_buf_ptr) 3375 return -EIO; 3376 3377 buf->host_buf_mem_type = WMFW_ADSP2_XM; 3378 3379 ret = wm_adsp_buffer_populate(buf); 3380 if (ret < 0) 3381 return ret; 3382 3383 adsp_dbg(dsp, "legacy host_buf_ptr=%x\n", buf->host_buf_ptr); 3384 3385 return 0; 3386 } 3387 3388 static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) 3389 { 3390 struct wm_adsp_host_buf_coeff_v1 coeff_v1; 3391 struct wm_adsp_compr_buf *buf; 3392 unsigned int val, reg; 3393 int ret, i; 3394 3395 ret = wm_coeff_base_reg(ctl, &reg); 3396 if (ret) 3397 return ret; 3398 3399 for (i = 0; i < 5; ++i) { 3400 ret = regmap_raw_read(ctl->dsp->regmap, reg, &val, sizeof(val)); 3401 if (ret < 0) 3402 return ret; 3403 3404 if (val) 3405 break; 3406 3407 usleep_range(1000, 2000); 3408 } 3409 3410 if (!val) { 3411 adsp_err(ctl->dsp, "Failed to acquire host buffer\n"); 3412 return -EIO; 3413 } 3414 3415 buf = wm_adsp_buffer_alloc(ctl->dsp); 3416 if (!buf) 3417 return -ENOMEM; 3418 3419 buf->host_buf_mem_type = ctl->alg_region.type; > 3420 buf->host_buf_ptr = be32_to_cpu(val); 3421 3422 ret = wm_adsp_buffer_populate(buf); 3423 if (ret < 0) 3424 return ret; 3425 3426 /* 3427 * v0 host_buffer coefficients didn't have versioning, so if the 3428 * control is one word, assume version 0. 3429 */ 3430 if (ctl->len == 4) { 3431 adsp_dbg(ctl->dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); 3432 return 0; 3433 } 3434 3435 ret = regmap_raw_read(ctl->dsp->regmap, reg, &coeff_v1, 3436 sizeof(coeff_v1)); 3437 if (ret < 0) 3438 return ret; 3439 > 3440 coeff_v1.versions = be32_to_cpu(coeff_v1.versions); > 3441 val = coeff_v1.versions & HOST_BUF_COEFF_COMPAT_VER_MASK; 3442 val >>= HOST_BUF_COEFF_COMPAT_VER_SHIFT; 3443 3444 if (val > HOST_BUF_COEFF_SUPPORTED_COMPAT_VER) { 3445 adsp_err(ctl->dsp, 3446 "Host buffer coeff ver %u > supported version %u\n", 3447 val, HOST_BUF_COEFF_SUPPORTED_COMPAT_VER); 3448 return -EINVAL; 3449 } 3450 3451 for (i = 0; i < ARRAY_SIZE(coeff_v1.name); i++) > 3452 coeff_v1.name[i] = be32_to_cpu(coeff_v1.name[i]); 3453 3454 wm_adsp_remove_padding((u32 *)&coeff_v1.name, 3455 ARRAY_SIZE(coeff_v1.name), 3456 WM_ADSP_DATA_WORD_SIZE); 3457 3458 buf->name = kasprintf(GFP_KERNEL, "%s-dsp-%s", ctl->dsp->part, 3459 (char *)&coeff_v1.name); 3460 3461 adsp_dbg(ctl->dsp, "host_buf_ptr=%x coeff version %u\n", 3462 buf->host_buf_ptr, val); 3463 3464 return val; 3465 } 3466 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 0/7] ASoC: stm32: i2s: miscellaneous fixes
by Olivier Moysan 26 Feb '19

26 Feb '19

This patch-set is a collection of fixes for STM32 I2S driver. Olivier Moysan (7): ASoC: stm32: i2s: fix IRQ clearing ASoC: stm32: i2s: fix 16 bit format support ASoC: stm32: i2s: fix stream count management ASoC: stm32: i2s: fix dma configuration ASoC: stm32: i2s: remove useless callback ASoC: stm32: i2s: fix race condition in irq handler ASoC: stm32: i2s: skip useless write in slave mode sound/soc/stm/stm32_i2s.c | 73 +++++++++++++++++++++++------------------------ 1 file changed, 35 insertions(+), 38 deletions(-) -- 2.7.4

2 14

[alsa-devel] [PATCH] ASoC: cs35l36: Fix an IS_ERR() vs NULL checking bug
by Dan Carpenter 26 Feb '19

26 Feb '19

The irq_get_irq_data() function doesn't return error pointers, it returns NULL. Fixes: 6ba9dd6c893b ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/cs35l36.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e374fffb7e17..901d57491218 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -1845,9 +1845,9 @@ static int cs35l36_i2c_probe(struct i2c_client *i2c_client, cs35l36_apply_vpbr_config(cs35l36); irq_d = irq_get_irq_data(i2c_client->irq); - if (IS_ERR(irq_d)) { + if (!irq_d) { dev_err(&i2c_client->dev, "Invalid IRQ: %d\n", i2c_client->irq); - ret = PTR_ERR(irq_d); + ret = -ENODEV; goto err; } -- 2.17.1

4 3

[alsa-devel] [PATCH 1/5] ASoC: wm_adsp: Factor out stripping padding from ADSP data
by Charles Keepax 26 Feb '19

26 Feb '19

In preparation for more refactoring add a helper function to strip the padding from ADSP data. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 8d16a14cd2266..ae7e1f5d8d10a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3231,6 +3231,21 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } +static void wm_adsp_remove_padding(u32 *buf, int nwords, int data_word_size) +{ + u8 *pack_in = (u8 *)buf; + u8 *pack_out = (u8 *)buf; + int i, j; + + /* Remove the padding bytes from the data read from the DSP */ + for (i = 0; i < nwords; i++) { + for (j = 0; j < data_word_size; j++) + *pack_out++ = *pack_in++; + + pack_in += sizeof(*buf) - data_word_size; + } +} + static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) { struct wm_adsp_alg_region *alg_region; @@ -3670,11 +3685,9 @@ EXPORT_SYMBOL_GPL(wm_adsp_compr_pointer); static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) { struct wm_adsp_compr_buf *buf = compr->buf; - u8 *pack_in = (u8 *)compr->raw_buf; - u8 *pack_out = (u8 *)compr->raw_buf; unsigned int adsp_addr; int mem_type, nwords, max_read; - int i, j, ret; + int i, ret; /* Calculate read parameters */ for (i = 0; i < wm_adsp_fw[buf->dsp->fw].caps->num_regions; ++i) @@ -3706,13 +3719,7 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) if (ret < 0) return ret; - /* Remove the padding bytes from the data read from the DSP */ - for (i = 0; i < nwords; i++) { - for (j = 0; j < WM_ADSP_DATA_WORD_SIZE; j++) - *pack_out++ = *pack_in++; - - pack_in += sizeof(*(compr->raw_buf)) - WM_ADSP_DATA_WORD_SIZE; - } + wm_adsp_remove_padding(compr->raw_buf, nwords, WM_ADSP_DATA_WORD_SIZE); /* update read index to account for words read */ buf->read_index += nwords; -- 2.11.0

2 8

[alsa-devel] Applied "ASoC: wm_adsp: Improve logging messages" to the asoc tree
by Mark Brown 26 Feb '19

26 Feb '19

The patch ASoC: wm_adsp: Improve logging messages 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 0d3fba3e7a566917f4286dd42b83c780c47dcbf7 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Fri, 22 Feb 2019 10:04:21 +0000 Subject: [PATCH] ASoC: wm_adsp: Improve logging messages As the compressed stream implementation has acquired support for multiple DAI links and compressed streams it has become harder to interpret messages in the kernel log. Add additional macros to include the compressed DAI name in the log messages, allowing different streams to be easily disambiguated. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 97 +++++++++++++++++++++----------------- 1 file changed, 53 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fe802fc331c5..8077c18cbcdf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -46,6 +46,13 @@ #define adsp_dbg(_dsp, fmt, ...) \ dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) +#define compr_err(_obj, fmt, ...) \ + adsp_err(_obj->dsp, "%s: " fmt, _obj->name ? _obj->name : "legacy", \ + ##__VA_ARGS__) +#define compr_dbg(_obj, fmt, ...) \ + adsp_dbg(_obj->dsp, "%s: " fmt, _obj->name ? _obj->name : "legacy", \ + ##__VA_ARGS__) + #define ADSP1_CONTROL_1 0x00 #define ADSP1_CONTROL_2 0x02 #define ADSP1_CONTROL_3 0x03 @@ -3033,20 +3040,23 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) mutex_lock(&dsp->pwr_lock); if (wm_adsp_fw[dsp->fw].num_caps == 0) { - adsp_err(dsp, "Firmware does not support compressed API\n"); + adsp_err(dsp, "%s: Firmware does not support compressed API\n", + rtd->codec_dai->name); ret = -ENXIO; goto out; } if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) { - adsp_err(dsp, "Firmware does not support stream direction\n"); + adsp_err(dsp, "%s: Firmware does not support stream direction\n", + rtd->codec_dai->name); ret = -EINVAL; goto out; } list_for_each_entry(tmp, &dsp->compr_list, list) { if (!strcmp(tmp->name, rtd->codec_dai->name)) { - adsp_err(dsp, "Only a single stream supported per dai\n"); + adsp_err(dsp, "%s: Only a single stream supported per dai\n", + rtd->codec_dai->name); ret = -EBUSY; goto out; } @@ -3106,9 +3116,9 @@ static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, params->buffer.fragments < WM_ADSP_MIN_FRAGMENTS || params->buffer.fragments > WM_ADSP_MAX_FRAGMENTS || params->buffer.fragment_size % WM_ADSP_DATA_WORD_SIZE) { - adsp_err(dsp, "Invalid buffer fragsize=%d fragments=%d\n", - params->buffer.fragment_size, - params->buffer.fragments); + compr_err(compr, "Invalid buffer fragsize=%d fragments=%d\n", + params->buffer.fragment_size, + params->buffer.fragments); return -EINVAL; } @@ -3136,9 +3146,9 @@ static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, return 0; } - adsp_err(dsp, "Invalid params id=%u ch=%u,%u rate=%u fmt=%u\n", - params->codec.id, params->codec.ch_in, params->codec.ch_out, - params->codec.sample_rate, params->codec.format); + compr_err(compr, "Invalid params id=%u ch=%u,%u rate=%u fmt=%u\n", + params->codec.id, params->codec.ch_in, params->codec.ch_out, + params->codec.sample_rate, params->codec.format); return -EINVAL; } @@ -3160,8 +3170,8 @@ int wm_adsp_compr_set_params(struct snd_compr_stream *stream, compr->size = params->buffer; - adsp_dbg(compr->dsp, "fragment_size=%d fragments=%d\n", - compr->size.fragment_size, compr->size.fragments); + compr_dbg(compr, "fragment_size=%d fragments=%d\n", + compr->size.fragment_size, compr->size.fragments); size = wm_adsp_compr_frag_words(compr) * sizeof(*compr->raw_buf); compr->raw_buf = kmalloc(size, GFP_DMA | GFP_KERNEL); @@ -3303,10 +3313,10 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; - adsp_dbg(buf->dsp, - "region=%d type=%d base=%08x off=%08x size=%08x\n", - i, region->mem_type, region->base_addr, - region->offset, region->cumulative_size); + compr_dbg(buf, + "region=%d type=%d base=%08x off=%08x size=%08x\n", + i, region->mem_type, region->base_addr, + region->offset, region->cumulative_size); } return 0; @@ -3380,7 +3390,7 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) if (ret < 0) return ret; - adsp_dbg(dsp, "legacy host_buf_ptr=%x\n", buf->host_buf_ptr); + compr_dbg(buf, "legacy host_buf_ptr=%x\n", buf->host_buf_ptr); return 0; } @@ -3428,7 +3438,7 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) * control is one word, assume version 0. */ if (ctl->len == 4) { - adsp_dbg(ctl->dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + compr_dbg(buf, "host_buf_ptr=%x\n", buf->host_buf_ptr); return 0; } @@ -3458,8 +3468,8 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) buf->name = kasprintf(GFP_KERNEL, "%s-dsp-%s", ctl->dsp->part, (char *)&coeff_v1.name); - adsp_dbg(ctl->dsp, "host_buf_ptr=%x coeff version %u\n", - buf->host_buf_ptr, val); + compr_dbg(buf, "host_buf_ptr=%x coeff version %u\n", + buf->host_buf_ptr, val); return val; } @@ -3525,7 +3535,7 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) struct wm_adsp *dsp = compr->dsp; int ret = 0; - adsp_dbg(dsp, "Trigger: %d\n", cmd); + compr_dbg(compr, "Trigger: %d\n", cmd); mutex_lock(&dsp->pwr_lock); @@ -3534,8 +3544,8 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) if (!wm_adsp_compr_attached(compr)) { ret = wm_adsp_compr_attach(compr); if (ret < 0) { - adsp_err(dsp, "Failed to link buffer and stream: %d\n", - ret); + compr_err(compr, "Failed to link buffer and stream: %d\n", + ret); break; } } @@ -3547,8 +3557,8 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) HOST_BUFFER_FIELD(high_water_mark), wm_adsp_compr_frag_words(compr)); if (ret < 0) { - adsp_err(dsp, "Failed to set high water mark: %d\n", - ret); + compr_err(compr, "Failed to set high water mark: %d\n", + ret); break; } break; @@ -3589,7 +3599,7 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) read_index = sign_extend32(next_read_index, 23); if (read_index < 0) { - adsp_dbg(buf->dsp, "Avail check on unstarted stream\n"); + compr_dbg(buf, "Avail check on unstarted stream\n"); return 0; } @@ -3607,8 +3617,8 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) if (avail < 0) avail += wm_adsp_buffer_size(buf); - adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); + compr_dbg(buf, "readindex=0x%x, writeindex=0x%x, avail=%d\n", + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; @@ -3621,11 +3631,11 @@ static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf) ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); if (ret < 0) { - adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret); + compr_err(buf, "Failed to check buffer error: %d\n", ret); return ret; } if (buf->error != 0) { - adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error); + compr_err(buf, "Buffer error occurred: %d\n", buf->error); return -EIO; } @@ -3644,6 +3654,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) ret = -ENODEV; goto out; } + adsp_dbg(dsp, "Handling buffer IRQ\n"); list_for_each_entry(buf, &dsp->buffer_list, list) { @@ -3656,13 +3667,13 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), &buf->irq_count); if (ret < 0) { - adsp_err(dsp, "Failed to get irq_count: %d\n", ret); + compr_err(buf, "Failed to get irq_count: %d\n", ret); goto out; } ret = wm_adsp_buffer_update_avail(buf); if (ret < 0) { - adsp_err(dsp, "Error reading avail: %d\n", ret); + compr_err(buf, "Error reading avail: %d\n", ret); goto out; } @@ -3686,8 +3697,7 @@ static int wm_adsp_buffer_reenable_irq(struct wm_adsp_compr_buf *buf) if (buf->irq_count & 0x01) return 0; - adsp_dbg(buf->dsp, "Enable IRQ(0x%x) for next fragment\n", - buf->irq_count); + compr_dbg(buf, "Enable IRQ(0x%x) for next fragment\n", buf->irq_count); buf->irq_count |= 0x01; @@ -3703,7 +3713,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct wm_adsp_compr_buf *buf; int ret = 0; - adsp_dbg(dsp, "Pointer request\n"); + compr_dbg(compr, "Pointer request\n"); mutex_lock(&dsp->pwr_lock); @@ -3718,7 +3728,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, if (buf->avail < wm_adsp_compr_frag_words(compr)) { ret = wm_adsp_buffer_update_avail(buf); if (ret < 0) { - adsp_err(dsp, "Error reading avail: %d\n", ret); + compr_err(compr, "Error reading avail: %d\n", ret); goto out; } @@ -3737,9 +3747,8 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, ret = wm_adsp_buffer_reenable_irq(buf); if (ret < 0) { - adsp_err(dsp, - "Failed to re-enable buffer IRQ: %d\n", - ret); + compr_err(compr, "Failed to re-enable buffer IRQ: %d\n", + ret); goto out; } } @@ -3814,11 +3823,10 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) static int wm_adsp_compr_read(struct wm_adsp_compr *compr, char __user *buf, size_t count) { - struct wm_adsp *dsp = compr->dsp; int ntotal = 0; int nwords, nbytes; - adsp_dbg(dsp, "Requested read of %zu bytes\n", count); + compr_dbg(compr, "Requested read of %zu bytes\n", count); if (!compr->buf || compr->buf->error) { snd_compr_stop_error(compr->stream, SNDRV_PCM_STATE_XRUN); @@ -3830,17 +3838,18 @@ static int wm_adsp_compr_read(struct wm_adsp_compr *compr, do { nwords = wm_adsp_buffer_capture_block(compr, count); if (nwords < 0) { - adsp_err(dsp, "Failed to capture block: %d\n", nwords); + compr_err(compr, "Failed to capture block: %d\n", + nwords); return nwords; } nbytes = nwords * WM_ADSP_DATA_WORD_SIZE; - adsp_dbg(dsp, "Read %d bytes\n", nbytes); + compr_dbg(compr, "Read %d bytes\n", nbytes); if (copy_to_user(buf + ntotal, compr->raw_buf, nbytes)) { - adsp_err(dsp, "Failed to copy data to user: %d, %d\n", - ntotal, nbytes); + compr_err(compr, "Failed to copy data to user: %d, %d\n", + ntotal, nbytes); return -EFAULT; } -- 2.20.1

1 0

[alsa-devel] [PATCH v2] ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
by twischer＠de.adit-jv.com 26 Feb '19

26 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif mode/adinr/dalign address is registered by: (in busif4 case) RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80) But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif mode/adinr/dalign register address ( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] ) are out of this rule. This patch registers ssi9 4/5/6/7 mode/adinr/dalign register as single register, and access these registers in case of SSI9 BUSIF 4/5/6/7. Fixes: commit 8c9d75033340 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Hi all, On 2/25/19 02:29, Kuninori Morimoto wrote: > I think > - and access these these registers > + and access these registers Done. On 2/25/19 02:29, Kuninori Morimoto wrote: > Necessary register on rsnd_mod_write() is just number today. > So how about this ? Code will be more simple/readable > > if ((id == 9) && (busif >= 4)) { > adinr = SSI9_BUSIF_ADINR(); > mode = SSI9_BUSIF_MODE(); > daligh = SSI9_BUSIF_DALIGN(); > } else { > adinr = SSI_BUSIF_ADINR(); > mode = SSI_BUSIF_MODE(); > daligh = SSI_BUSIF_DALIGN(); > } > > rsnd_mod_write(mod, adinr, > rsnd_get_adinr_bit(mod, io) | chnl); > rsnd_mod_write(mod, mode, > rsnd_get_busif_shift(io, mod) | 1); > rsnd_mod_write(mod, dalign, > rsnd_get_dalign(mod, io)); Done. Best regards Timo sound/soc/sh/rcar/gen.c | 24 ++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 27 +++++++++++++++++++++++++++ sound/soc/sh/rcar/ssiu.c | 24 +++++++++++------------- 3 files changed, 62 insertions(+), 13 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 7cda601..af19010 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -255,6 +255,30 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c), + RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484), + RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488), + RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0), + RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4), + RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8), + RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0), + RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4), + RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8), + RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0), + RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4), + RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8), + RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80), + RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84), + RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88), + RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0), + RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4), + RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8), + RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0), + RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4), + RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8), + RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0), + RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4), + RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8), }; static const struct rsnd_regmap_field_conf conf_scu[] = { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 605e4b9..90625c5 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -191,6 +191,30 @@ enum rsnd_reg { SSI_SYS_STATUS7, HDMI0_SEL, HDMI1_SEL, + SSI9_BUSIF0_MODE, + SSI9_BUSIF1_MODE, + SSI9_BUSIF2_MODE, + SSI9_BUSIF3_MODE, + SSI9_BUSIF4_MODE, + SSI9_BUSIF5_MODE, + SSI9_BUSIF6_MODE, + SSI9_BUSIF7_MODE, + SSI9_BUSIF0_ADINR, + SSI9_BUSIF1_ADINR, + SSI9_BUSIF2_ADINR, + SSI9_BUSIF3_ADINR, + SSI9_BUSIF4_ADINR, + SSI9_BUSIF5_ADINR, + SSI9_BUSIF6_ADINR, + SSI9_BUSIF7_ADINR, + SSI9_BUSIF0_DALIGN, + SSI9_BUSIF1_DALIGN, + SSI9_BUSIF2_DALIGN, + SSI9_BUSIF3_DALIGN, + SSI9_BUSIF4_DALIGN, + SSI9_BUSIF5_DALIGN, + SSI9_BUSIF6_DALIGN, + SSI9_BUSIF7_DALIGN, /* SSI */ SSICR, @@ -209,6 +233,9 @@ enum rsnd_reg { #define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) #define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) #define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI9_BUSIF_MODE(i) (SSI9_BUSIF0_MODE + (i)) +#define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) +#define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index c74991d..2347f34 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -181,28 +181,26 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); int busif = rsnd_mod_id_sub(mod); + enum rsnd_reg adinr_reg, mode_reg, dalign_reg; - /* - * FIXME - * - * We can't support SSI9-4/5/6/7, because its address is - * out of calculation rule - */ if ((id == 9) && (busif >= 4)) { - struct device *dev = rsnd_priv_to_dev(priv); - - dev_err(dev, "This driver doesn't support SSI%d-%d, so far", - id, busif); + adinr_reg = SSI9_BUSIF_ADINR(busif); + mode_reg = SSI9_BUSIF_MODE(busif); + dalign_reg = SSI9_BUSIF_DALIGN(busif); + } else { + adinr_reg = SSI_BUSIF_ADINR(busif); + mode_reg = SSI_BUSIF_MODE(busif); + dalign_reg = SSI_BUSIF_DALIGN(busif); } - rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), + rsnd_mod_write(mod, adinr_reg, rsnd_get_adinr_bit(mod, io) | (rsnd_io_is_play(io) ? rsnd_runtime_channel_after_ctu(io) : rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), + rsnd_mod_write(mod, mode_reg, rsnd_get_busif_shift(io, mod) | 1); - rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), + rsnd_mod_write(mod, dalign_reg, rsnd_get_dalign(mod, io)); } -- 2.7.4

3 2

[alsa-devel] Applied "ASoC: Intel: Boards: Add Maxim98373 support" to the asoc tree
by Mark Brown 26 Feb '19

26 Feb '19

The patch ASoC: Intel: Boards: Add Maxim98373 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 716d53cc7837aec7f439ce2a20fc2597a89dae53 Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Mon, 25 Feb 2019 22:17:31 +0530 Subject: [PATCH] ASoC: Intel: Boards: Add Maxim98373 support This patch enables the reuse of kbl_da7219_max98927 machine driver to support max98373. The same machine driver is modified for cases where one amplifier is swapped out with another. Most of the changes are about renaming the codec and codec_dai names, with minor differences due to support for 24 bits in one case and 16 in the other. 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/boards/Kconfig | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 203 ++++++++++++++++-- .../intel/common/soc-acpi-intel-kbl-match.c | 19 ++ 3 files changed, 200 insertions(+), 23 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 0a7e40d06395..12d6b73e9531 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,6 +293,7 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_DA7219 select SND_SOC_MAX98927 + select SND_SOC_MAX98373 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 6dd5c69671b3..2768a572d065 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -2,7 +2,7 @@ // Copyright(c) 2018 Intel Corporation. /* - * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs + * Intel Kabylake I2S Machine Driver with MAX98927, MAX98373 & DA7219 Codecs * * Modified from: * Intel Kabylake I2S Machine driver supporting MAX98927 and @@ -24,8 +24,14 @@ #define KBL_DIALOG_CODEC_DAI "da7219-hifi" #define MAX98927_CODEC_DAI "max98927-aif1" -#define MAXIM_DEV0_NAME "i2c-MX98927:00" -#define MAXIM_DEV1_NAME "i2c-MX98927:01" +#define MAX98927_DEV0_NAME "i2c-MX98927:00" +#define MAX98927_DEV1_NAME "i2c-MX98927:01" + +#define MAX98373_CODEC_DAI "max98373-aif1" +#define MAX98373_DEV0_NAME "i2c-MX98373:00" +#define MAX98373_DEV1_NAME "i2c-MX98373:01" + + #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 #define NAME_SIZE 32 @@ -176,20 +182,38 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, for (j = 0; j < runtime->num_codecs; j++) { struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; - if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); return ret; } } - if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + if (!strcmp(codec_dai->component->name, MAX98927_DEV1_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); return ret; } } + if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x03, 3, 8, 24); + if (ret < 0) { + dev_err(runtime->dev, + "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x0C, 3, 8, 24); + if (ret < 0) { + dev_err(runtime->dev, + "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } } return 0; @@ -212,6 +236,25 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + /* + * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE, + * where as kblda7219m98927 & kblmax98927 supports S16_LE by default. + * Skipping the port wise FE and BE configuration for kblda7219m98373 & + * kblmax98373 as the topology (FE & BE) supports S24_LE only. + */ + + if (!strcmp(rtd->card->name, "kblda7219m98373") || + !strcmp(rtd->card->name, "kblmax98373")) { + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; + } + /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ @@ -352,20 +395,31 @@ static struct snd_pcm_hw_constraint_list constraints_channels_quad = { static int kbl_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_rt = substream->private_data; /* * On this platform for PCM device we support, * 48Khz * stereo - * 16 bit audio */ runtime->hw.channels_max = DUAL_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + /* + * Setup S24_LE (32 bit container and 24 bit valid data) for + * kblda7219m98373 & kblmax98373. For kblda7219m98927 & + * kblmax98927 keeping it as 16/16 due to topology FW dependency. + */ + if (!strcmp(soc_rt->card->name, "kblda7219m98373") || + !strcmp(soc_rt->card->name, "kblmax98373")) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + + } else { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + } snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); @@ -398,11 +452,23 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, static int kabylake_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_rt = substream->private_data; runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels_quad); + /* + * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE. + * The DMIC also configured for S24_LE. Forcing the DMIC format to + * S24_LE due to the topology FW dependency. + */ + if (!strcmp(soc_rt->card->name, "kblda7219m98373") || + !strcmp(soc_rt->card->name, "kblmax98373")) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + } + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } @@ -448,29 +514,55 @@ static struct snd_soc_ops skylake_refcap_ops = { static struct snd_soc_codec_conf max98927_codec_conf[] = { { - .dev_name = MAXIM_DEV0_NAME, + .dev_name = MAX98927_DEV0_NAME, + .name_prefix = "Right", + }, + + { + .dev_name = MAX98927_DEV1_NAME, + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max98373_codec_conf[] = { + + { + .dev_name = MAX98373_DEV0_NAME, .name_prefix = "Right", }, { - .dev_name = MAXIM_DEV1_NAME, + .dev_name = MAX98373_DEV1_NAME, .name_prefix = "Left", }, }; -static struct snd_soc_dai_link_component ssp0_codec_components[] = { +static struct snd_soc_dai_link_component max98927_ssp0_codec_components[] = { { /* Left */ - .name = MAXIM_DEV0_NAME, + .name = MAX98927_DEV0_NAME, .dai_name = MAX98927_CODEC_DAI, }, { /* For Right */ - .name = MAXIM_DEV1_NAME, + .name = MAX98927_DEV1_NAME, .dai_name = MAX98927_CODEC_DAI, }, }; +static struct snd_soc_dai_link_component max98373_ssp0_codec_components[] = { + { /* Left */ + .name = MAX98373_DEV0_NAME, + .dai_name = MAX98373_CODEC_DAI, + }, + + { /* For Right */ + .name = MAX98373_DEV1_NAME, + .dai_name = MAX98373_CODEC_DAI, + }, + +}; + /* kabylake digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link kabylake_dais[] = { /* Front End DAI links */ @@ -607,8 +699,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, - .codecs = ssp0_codec_components, - .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .codecs = max98927_ssp0_codec_components, + .num_codecs = ARRAY_SIZE(max98927_ssp0_codec_components), .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -683,7 +775,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { }; /* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_max98927_dais[] = { +static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { /* Front End DAI links */ [KBL_DPCM_AUDIO_PB] = { .name = "Kbl Audio Port", @@ -802,8 +894,8 @@ static struct snd_soc_dai_link kabylake_max98927_dais[] = { .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, - .codecs = ssp0_codec_components, - .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .codecs = max98927_ssp0_codec_components, + .num_codecs = ARRAY_SIZE(max98927_ssp0_codec_components), .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -917,8 +1009,8 @@ static struct snd_soc_card kbl_audio_card_da7219_m98927 = { static struct snd_soc_card kbl_audio_card_max98927 = { .name = "kblmax98927", .owner = THIS_MODULE, - .dai_link = kabylake_max98927_dais, - .num_links = ARRAY_SIZE(kabylake_max98927_dais), + .dai_link = kabylake_max98_927_373_dais, + .num_links = ARRAY_SIZE(kabylake_max98_927_373_dais), .controls = kabylake_controls, .num_controls = ARRAY_SIZE(kabylake_controls), .dapm_widgets = kabylake_widgets, @@ -931,9 +1023,46 @@ static struct snd_soc_card kbl_audio_card_max98927 = { .late_probe = kabylake_card_late_probe, }; +static struct snd_soc_card kbl_audio_card_da7219_m98373 = { + .name = "kblda7219m98373", + .owner = THIS_MODULE, + .dai_link = kabylake_dais, + .num_links = ARRAY_SIZE(kabylake_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98373_codec_conf, + .num_configs = ARRAY_SIZE(max98373_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static struct snd_soc_card kbl_audio_card_max98373 = { + .name = "kblmax98373", + .owner = THIS_MODULE, + .dai_link = kabylake_max98_927_373_dais, + .num_links = ARRAY_SIZE(kabylake_max98_927_373_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98373_codec_conf, + .num_configs = ARRAY_SIZE(max98373_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; + struct snd_soc_dai_link *kbl_dai_link; + struct snd_soc_dai_link_component **codecs; + int i = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -944,6 +1073,22 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card = (struct snd_soc_card *)pdev->id_entry->driver_data; + kbl_dai_link = kabylake_audio_card->dai_link; + + /* Update codecs for SSP0 with max98373 codec info */ + if (!strcmp(pdev->name, "kbl_da7219_max98373") || + (!strcmp(pdev->name, "kbl_max98373"))) { + for (i = 0; i < kabylake_audio_card->num_links; ++i) { + if (strcmp(kbl_dai_link[i].name, "SSP0-Codec")) + continue; + + codecs = &(kbl_dai_link[i].codecs); + *codecs = max98373_ssp0_codec_components; + kbl_dai_link[i].num_codecs = + ARRAY_SIZE(max98373_ssp0_codec_components); + break; + } + } kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); @@ -961,13 +1106,23 @@ static const struct platform_device_id kbl_board_ids[] = { .driver_data = (kernel_ulong_t)&kbl_audio_card_max98927, }, + { + .name = "kbl_da7219_max98373", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_da7219_m98373, + }, + { + .name = "kbl_max98373", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_max98373, + }, { } }; static struct platform_driver kabylake_audio = { .probe = kabylake_audio_probe, .driver = { - .name = "kbl_da7219_max98927", + .name = "kbl_da7219_max98_927_373", .pm = &snd_soc_pm_ops, }, .id_table = kbl_board_ids, @@ -976,8 +1131,10 @@ static struct platform_driver kabylake_audio = { module_platform_driver(kabylake_audio) /* Module information */ -MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219"); +MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927/MAX98373 & DA7219"); MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:kbl_da7219_max98927"); MODULE_ALIAS("platform:kbl_max98927"); +MODULE_ALIAS("platform:kbl_da7219_max98373"); +MODULE_ALIAS("platform:kbl_max98373"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index e6fa6f470526..4b331058e807 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -37,6 +37,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = { .codecs = {"MX98927"} }; +static struct snd_soc_acpi_codecs kbl_7219_98373_codecs = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { { .id = "INT343A", @@ -106,6 +111,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .drv_name = "kbl_rt5660", .fw_filename = "intel/dsp_fw_kbl.bin", }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98373", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98373_codecs, + .pdata = &skl_dmic_data + }, + { + .id = "MX98373", + .drv_name = "kbl_max98373", + .fw_filename = "intel/dsp_fw_kbl.bin", + .pdata = &skl_dmic_data + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: qcom: Kconfig: fix dependency for sdm845
by Cheng-Yi Chiang 26 Feb '19

26 Feb '19

SND_SOC_CROS_EC_CODEC depends on MFD_CROS_EC. Add that dependency to SND_SOC_SDM845 to fix unmet direct dependencies warning. Fixes: 74c6ecf4194e (ASoC: qcom: Kconfig: select dmic for sdm845) Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 8f206cb4fcc0..75ceb04d8bf0 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -98,7 +98,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR + depends on QCOM_APR && MFD_CROS_EC select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 -- 2.21.0.rc0.258.g878e2cd30e-goog

4 3

[alsa-devel] [PATCH v2] arm64: dts: renesas: r8a77965: add SSIU support for sound
by twischer＠de.adit-jv.com 26 Feb '19

26 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> rsnd driver supports SSIU now, let's use it. Then, BUSIF DMA settings on rcar_sound,ssi (= rxu, txu) are no longer needed. Applies commit 8d14bfa074db ("arm64: dts: renesas: r8a7796: add SSIU support for sound") and commit 10bd03fa896e ("arm64: dts: renesas: r8a7796: remove BUSIF0 settings from rcar_sound,ssi") for r8a77965. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Hi all, On 2/25/19 02:41, Kuninori Morimoto wrote: > I think ARM maintainer doesn't want duplicate/copied patches. > So, we want/need to merge into this patch, too > > 10bd03fa896ee5e25fa3548a3d1112b9beb76943 > ("arm64: dts: renesas: r8a7796: remove BUSIF0 settings from rcar_sound,ssi") Done. > And I guess this patch is related, so, we need to merge it too ? > > arm64: dts: renesas: r8a7796: use extended audio dmac register There are some ongoing discussions. I would expect some more delay for this patch. This patch is already helpful because BUSIF0-4 can be used without: arm64: dts: renesas: r8a7796: use extended audio dmac register Therefore I would like to not block this patch by arm64: dts: renesas: r8a7796: use extended audio dmac register Best regards Timo arch/arm64/boot/dts/renesas/r8a77965.dtsi | 251 +++++++++++++++++++++++++++--- 1 file changed, 231 insertions(+), 20 deletions(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index 4b3730f..7c081d6 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1585,56 +1585,267 @@ }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + ssiu02: ssiu-2 { + dmas = <&audma0 0x37>, <&audma1 0x38>; + dma-names = "rx", "tx"; + }; + ssiu03: ssiu-3 { + dmas = <&audma0 0x47>, <&audma1 0x48>; + dma-names = "rx", "tx"; + }; + ssiu04: ssiu-4 { + dmas = <&audma0 0x3F>, <&audma1 0x40>; + dma-names = "rx", "tx"; + }; + ssiu05: ssiu-5 { + dmas = <&audma0 0x43>, <&audma1 0x44>; + dma-names = "rx", "tx"; + }; + ssiu06: ssiu-6 { + dmas = <&audma0 0x4F>, <&audma1 0x50>; + dma-names = "rx", "tx"; + }; + ssiu07: ssiu-7 { + dmas = <&audma0 0x53>, <&audma1 0x54>; + dma-names = "rx", "tx"; + }; + ssiu10: ssiu-8 { + dmas = <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx"; + }; + ssiu11: ssiu-9 { + dmas = <&audma0 0x4B>, <&audma1 0x4C>; + dma-names = "rx", "tx"; + }; + ssiu12: ssiu-10 { + dmas = <&audma0 0x57>, <&audma1 0x58>; + dma-names = "rx", "tx"; + }; + ssiu13: ssiu-11 { + dmas = <&audma0 0x59>, <&audma1 0x5A>; + dma-names = "rx", "tx"; + }; + ssiu14: ssiu-12 { + dmas = <&audma0 0x5F>, <&audma1 0x60>; + dma-names = "rx", "tx"; + }; + ssiu15: ssiu-13 { + dmas = <&audma0 0xC3>, <&audma1 0xC4>; + dma-names = "rx", "tx"; + }; + ssiu16: ssiu-14 { + dmas = <&audma0 0xC7>, <&audma1 0xC8>; + dma-names = "rx", "tx"; + }; + ssiu17: ssiu-15 { + dmas = <&audma0 0xCB>, <&audma1 0xCC>; + dma-names = "rx", "tx"; + }; + ssiu20: ssiu-16 { + dmas = <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx"; + }; + ssiu21: ssiu-17 { + dmas = <&audma0 0x67>, <&audma1 0x68>; + dma-names = "rx", "tx"; + }; + ssiu22: ssiu-18 { + dmas = <&audma0 0x6B>, <&audma1 0x6C>; + dma-names = "rx", "tx"; + }; + ssiu23: ssiu-19 { + dmas = <&audma0 0x6D>, <&audma1 0x6E>; + dma-names = "rx", "tx"; + }; + ssiu24: ssiu-20 { + dmas = <&audma0 0xCF>, <&audma1 0xCE>; + dma-names = "rx", "tx"; + }; + ssiu25: ssiu-21 { + dmas = <&audma0 0xEB>, <&audma1 0xEC>; + dma-names = "rx", "tx"; + }; + ssiu26: ssiu-22 { + dmas = <&audma0 0xED>, <&audma1 0xEE>; + dma-names = "rx", "tx"; + }; + ssiu27: ssiu-23 { + dmas = <&audma0 0xEF>, <&audma1 0xF0>; + dma-names = "rx", "tx"; + }; + ssiu30: ssiu-24 { + dmas = <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx"; + }; + ssiu31: ssiu-25 { + dmas = <&audma0 0x21>, <&audma1 0x22>; + dma-names = "rx", "tx"; + }; + ssiu32: ssiu-26 { + dmas = <&audma0 0x23>, <&audma1 0x24>; + dma-names = "rx", "tx"; + }; + ssiu33: ssiu-27 { + dmas = <&audma0 0x25>, <&audma1 0x26>; + dma-names = "rx", "tx"; + }; + ssiu34: ssiu-28 { + dmas = <&audma0 0x27>, <&audma1 0x28>; + dma-names = "rx", "tx"; + }; + ssiu35: ssiu-29 { + dmas = <&audma0 0x29>, <&audma1 0x2A>; + dma-names = "rx", "tx"; + }; + ssiu36: ssiu-30 { + dmas = <&audma0 0x2B>, <&audma1 0x2C>; + dma-names = "rx", "tx"; + }; + ssiu37: ssiu-31 { + dmas = <&audma0 0x2D>, <&audma1 0x2E>; + dma-names = "rx", "tx"; + }; + ssiu40: ssiu-32 { + dmas = <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx"; + }; + ssiu41: ssiu-33 { + dmas = <&audma0 0x17>, <&audma1 0x18>; + dma-names = "rx", "tx"; + }; + ssiu42: ssiu-34 { + dmas = <&audma0 0x19>, <&audma1 0x1A>; + dma-names = "rx", "tx"; + }; + ssiu43: ssiu-35 { + dmas = <&audma0 0x1B>, <&audma1 0x1C>; + dma-names = "rx", "tx"; + }; + ssiu44: ssiu-36 { + dmas = <&audma0 0x1D>, <&audma1 0x1E>; + dma-names = "rx", "tx"; + }; + ssiu45: ssiu-37 { + dmas = <&audma0 0x1F>, <&audma1 0x20>; + dma-names = "rx", "tx"; + }; + ssiu46: ssiu-38 { + dmas = <&audma0 0x31>, <&audma1 0x32>; + dma-names = "rx", "tx"; + }; + ssiu47: ssiu-39 { + dmas = <&audma0 0x33>, <&audma1 0x34>; + dma-names = "rx", "tx"; + }; + ssiu50: ssiu-40 { + dmas = <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx"; + }; + ssiu60: ssiu-41 { + dmas = <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx"; + }; + ssiu70: ssiu-42 { + dmas = <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx"; + }; + ssiu80: ssiu-43 { + dmas = <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx"; + }; + ssiu90: ssiu-44 { + dmas = <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx"; + }; + ssiu91: ssiu-45 { + dmas = <&audma0 0x7F>, <&audma1 0x80>; + dma-names = "rx", "tx"; + }; + ssiu92: ssiu-46 { + dmas = <&audma0 0x81>, <&audma1 0x82>; + dma-names = "rx", "tx"; + }; + ssiu93: ssiu-47 { + dmas = <&audma0 0x83>, <&audma1 0x84>; + dma-names = "rx", "tx"; + }; + ssiu94: ssiu-48 { + dmas = <&audma0 0xA3>, <&audma1 0xA4>; + dma-names = "rx", "tx"; + }; + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; }; ssi1: ssi-1 { interrupts = <GIC_SPI 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; }; ssi2: ssi-2 { interrupts = <GIC_SPI 372 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x05>, <&audma1 0x06>; + dma-names = "rx", "tx"; }; ssi3: ssi-3 { interrupts = <GIC_SPI 373 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x07>, <&audma1 0x08>; + dma-names = "rx", "tx"; }; ssi4: ssi-4 { interrupts = <GIC_SPI 374 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x09>, <&audma1 0x0a>; + dma-names = "rx", "tx"; }; ssi5: ssi-5 { interrupts = <GIC_SPI 375 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x0b>, <&audma1 0x0c>; + dma-names = "rx", "tx"; }; ssi6: ssi-6 { interrupts = <GIC_SPI 376 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x0d>, <&audma1 0x0e>; + dma-names = "rx", "tx"; }; ssi7: ssi-7 { interrupts = <GIC_SPI 377 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x0f>, <&audma1 0x10>; + dma-names = "rx", "tx"; }; ssi8: ssi-8 { interrupts = <GIC_SPI 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x11>, <&audma1 0x12>; + dma-names = "rx", "tx"; }; ssi9: ssi-9 { interrupts = <GIC_SPI 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x13>, <&audma1 0x14>; + dma-names = "rx", "tx"; }; }; }; -- 2.7.4

3 2

[alsa-devel] [PATCH] ALSA: firewire-motu: fix construction of PCM frame for capture direction
by Takashi Sakamoto 26 Feb '19

26 Feb '19

In data blocks of common isochronous packet for MOTU devices, PCM frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described in IEC 61883-6. The frames are not aligned to quadlet. For capture PCM substream, ALSA firewire-motu driver constructs PCM frames by reading data blocks byte-by-byte. However this operation includes bug for lower byte of the PCM sample. This brings invalid content of the PCM samples. This commit fixes the bug. Reported-by: Peter Sjöberg <autopeter(a)gmail.com> Cc: <stable(a)vger.kernel.org> # v4.12+ Fixes: 4641c9394010 ("ALSA: firewire-motu: add MOTU specific protocol layer") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index f0555a24d90e..6c9b743ea74b 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -136,7 +136,9 @@ static void read_pcm_s32(struct amdtp_stream *s, byte = (u8 *)buffer + p->pcm_byte_offset; for (c = 0; c < channels; ++c) { - *dst = (byte[0] << 24) | (byte[1] << 16) | byte[2]; + *dst = (byte[0] << 24) | + (byte[1] << 16) | + (byte[2] << 8); byte += 3; dst++; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against Liquid Saffire 56
by Takashi Sakamoto 26 Feb '19

26 Feb '19

ALSA bebob driver has an entry for Focusrite Saffire Pro 10 I/O. The entry matches vendor_id in root directory and model_id in unit directory of configuration ROM for IEEE 1394 bus. On the other hand, configuration ROM of Focusrite Liquid Saffire 56 has the same vendor_id and model_id. This device is an application of TCAT Dice (TCD2220 a.k.a Dice Jr.) however ALSA bebob driver can be bound to it randomly instead of ALSA dice driver. At present, drivers in ALSA firewire stack can not handle this situation appropriately. This commit uses more identical mod_alias for Focusrite Saffire Pro 10 I/O in ALSA bebob driver. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 042a829d bus_info_length 4, crc_length 42, crc 33437 404 31333934 bus_name "1394" 408 f0649222 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100, max_rec 9 (1024), max_rom 2, gen 2, spd 2 (S400) 40c 00130e01 company_id 00130e | 410 000606e0 device_id 01000606e0 | EUI-64 00130e01000606e0 root directory ----------------------------------------------------------------- 414 0009d31c directory_length 9, crc 54044 418 04000014 hardware version 41c 0c0083c0 node capabilities per IEEE 1394 420 0300130e vendor 424 81000012 --> descriptor leaf at 46c 428 17000006 model 42c 81000016 --> descriptor leaf at 484 430 130120c2 version 434 d1000002 --> unit directory at 43c 438 d4000006 --> dependent info directory at 450 unit directory at 43c ----------------------------------------------------------------- 43c 0004707c directory_length 4, crc 28796 440 1200a02d specifier id: 1394 TA 444 13010001 version: AV/C 448 17000006 model 44c 81000013 --> descriptor leaf at 498 dependent info directory at 450 ----------------------------------------------------------------- 450 000637c7 directory_length 6, crc 14279 454 120007f5 specifier id 458 13000001 version 45c 3affffc7 (immediate value) 460 3b100000 (immediate value) 464 3cffffc7 (immediate value) 468 3d600000 (immediate value) descriptor leaf at 46c ----------------------------------------------------------------- 46c 00056f3b leaf_length 5, crc 28475 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 466f6375 "Focu" 47c 73726974 "srit" 480 65000000 "e" descriptor leaf at 484 ----------------------------------------------------------------- 484 0004a165 leaf_length 4, crc 41317 488 00000000 textual descriptor 48c 00000000 minimal ASCII 490 50726f31 "Pro1" 494 30494f00 "0IO" descriptor leaf at 498 ----------------------------------------------------------------- 498 0004a165 leaf_length 4, crc 41317 49c 00000000 textual descriptor 4a0 00000000 minimal ASCII 4a4 50726f31 "Pro1" 4a8 30494f00 "0IO" $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 040442e4 bus_info_length 4, crc_length 4, crc 17124 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 00130e04 company_id 00130e | 410 018001e9 device_id 04018001e9 | EUI-64 00130e04018001e9 root directory ----------------------------------------------------------------- 414 00065612 directory_length 6, crc 22034 418 0300130e vendor 41c 8100000a --> descriptor leaf at 444 420 17000006 model 424 8100000e --> descriptor leaf at 45c 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 000418a0 directory_length 4, crc 6304 434 1200130e specifier id 438 13000001 version 43c 17000006 model 440 8100000f --> descriptor leaf at 47c descriptor leaf at 444 ----------------------------------------------------------------- 444 00056f3b leaf_length 5, crc 28475 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 466f6375 "Focu" 454 73726974 "srit" 458 65000000 "e" descriptor leaf at 45c ----------------------------------------------------------------- 45c 000762c6 leaf_length 7, crc 25286 460 00000000 textual descriptor 464 00000000 minimal ASCII 468 4c495155 "LIQU" 46c 49445f53 "ID_S" 470 41464649 "AFFI" 474 52455f35 "RE_5" 478 36000000 "6" descriptor leaf at 47c ----------------------------------------------------------------- 47c 000762c6 leaf_length 7, crc 25286 480 00000000 textual descriptor 484 00000000 minimal ASCII 488 4c495155 "LIQU" 48c 49445f53 "ID_S" 490 41464649 "AFFI" 494 52455f35 "RE_5" 498 36000000 "6" Cc: <stable(a)vger.kernel.org> # v3.16+ Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index d91874275d2c..5b46e8dcc2dd 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -448,7 +448,19 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Focusrite, SaffirePro 26 I/O */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec), /* Focusrite, SaffirePro 10 I/O */ - SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec), + { + // The combination of vendor_id and model_id is the same as the + // same as the one of Liquid Saffire 56. + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VEN_FOCUSRITE, + .model_id = 0x000006, + .specifier_id = 0x00a02d, + .version = 0x010001, + .driver_data = (kernel_ulong_t)&saffirepro_10_spec, + }, /* Focusrite, Saffire(no label and LE) */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH, &saffire_spec), -- 2.19.1

2 1

[alsa-devel] [PATCH V2] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 26 Feb '19

26 Feb '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 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 - update commit comments. sound/soc/fsl/fsl_esai.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..23bd0ad4ac31 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: @@ -611,12 +606,23 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, 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 +912,12 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + /* 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 4

[alsa-devel] [PATCH] ASoC:hdac_hda: use correct format to setup hda codec
by Rander Wang 25 Feb '19

25 Feb '19

The current implementation of the hdac_hda codec results in zero-valued samples on capture and noise with headset playback when SOF is used on platforms with an on-board HDaudio codec. This is root-caused to SOF using be_hw_params_fixup, and the prepare() call using invalid runtime fields to determine the format. This patch moves the format handling to the hw_params() callback, as done already for hdac_hdmi, to make sure the fixed-up information is taken into account but keeps the codec initialization in prepare() as the stream_tag is only available at that time. Moving everything in the prepare() callback is possible but the code is less elegant so this two-step solution was chosen. The solution was tested with the SST driver with no regressions, and all the issues with SOF playback and capture are solved. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 53 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/hdac_hda.h | 1 + 2 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index ffecdaaa8cf2..f889d94c8e3c 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -38,6 +38,9 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -50,6 +53,7 @@ static const struct snd_soc_dai_ops hdac_hda_dai_ops = { .startup = hdac_hda_dai_open, .shutdown = hdac_hda_dai_close, .prepare = hdac_hda_dai_prepare, + .hw_params = hdac_hda_dai_hw_params, .hw_free = hdac_hda_dai_hw_free, .set_tdm_slot = hdac_hda_dai_set_tdm_slot, }; @@ -139,6 +143,39 @@ static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int hdac_hda_dai_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 hdac_hda_priv *hda_pvt; + unsigned int format_val; + unsigned int maxbps; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + maxbps = dai->driver->playback.sig_bits; + else + maxbps = dai->driver->capture.sig_bits; + + hda_pvt = snd_soc_component_get_drvdata(component); + format_val = snd_hdac_calc_stream_format(params_rate(params), + params_channels(params), + params_format(params), + maxbps, + 0); + if (!format_val) { + dev_err(dai->dev, + "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + params_rate(params), params_channels(params), + params_format(params), maxbps); + + return -EINVAL; + } + + hda_pvt->pcm[dai->id].format_val[substream->stream] = format_val; + return 0; +} + static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,10 +199,9 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct snd_pcm_runtime *runtime = substream->runtime; struct hdac_device *hdev; - struct hda_pcm_stream *hda_stream; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; @@ -179,19 +215,8 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hda_stream->maxbps, - 0); - if (!format_val) { - dev_err(&hdev->dev, - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - return -EINVAL; - } - stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, stream, format_val, substream); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index e444ef593360..6b1bd4f428e7 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -8,6 +8,7 @@ struct hdac_hda_pcm { int stream_tag[2]; + unsigned int format_val[2]; }; struct hdac_hda_priv { -- 2.14.1

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[alsa-devel] Merge window for 5.1
by Takashi Iwai 25 Feb '19

25 Feb '19

Hi, I'll be traveling in the whole next week (from March 3 to 11), and likely offline during that. This schedule will conflict with the first week of the planned 5.1 merge window, so I'd like to close the merge a bit earlier than usual at this time. That is, if anyone has some urgent patch to be merged for 5.1, please submit ASAP for review. Mark, could you prepare the ASoC merge materials until Thursday? Then I'll send an early pull request to Linus on the upcoming weekend. Small fixes can be covered in the second week of the merge window, we should push out the major changes beforehand. thanks, Takashi

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Boards: Add Maxim98373 support
by Jenny TC 25 Feb '19

25 Feb '19

Reuse kbl_da7219_max98927 machine driver to support max98373 v1: First patchset V2: Updated commit message and Acked-by Jenny TC (1): ASoC: Intel: Boards: Add Maxim98373 support sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 203 +++++++++++++++++++--- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 19 ++ 3 files changed, 200 insertions(+), 23 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda: Extend i915 component bind timeout
by Takashi Iwai 25 Feb '19

25 Feb '19

I set 10 seconds for the timeout of the i915 audio component binding with a hope that recent machines are fast enough to handle all probe tasks in that period, but I was too optimistic. The binding may take longer than that, and this caused a problem on the machine with both audio and graphics driver modules loaded in parallel, as Paul Menzel experienced. This problem haven't hit so often just because the KMS driver is loaded in initrd on most machines. As a simple workaround, extend the timeout to 60 seconds. Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding") Reported-by: Paul Menzel <pmenzel+alsa-devel(a)molgen.mpg.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 617ff1aa818f..27eb0270a711 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -144,9 +144,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) return -ENODEV; if (!acomp->ops) { request_module("i915"); - /* 10s timeout */ + /* 60s timeout */ wait_for_completion_timeout(&bind_complete, - msecs_to_jiffies(10 * 1000)); + msecs_to_jiffies(60 * 1000)); } if (!acomp->ops) { dev_info(bus->dev, "couldn't bind with audio component\n"); -- 2.16.4

1 0

[alsa-devel] A bug in the alsa project
by Радослав Ненчовски 25 Feb '19

25 Feb '19

*Hi. I think I found a bug in the ALSA project which doesn't seem to be kernel related, since it occurs with all kernels I've tried so far. It's about the area where one can select the output - Stereo, 2.1, 4.0 and so on. I have a 2.1 audio system - 2 speakers and a subwoofer (Microlab M-300U). If I choose 5.1 Analog Output, the subwoofer works great and the audio has great bass for music. However, if I choose anything else that is not 5.1 (that includes the output for my system - 2.1), the subwoofer stops working and the output sounds like I'm using cheap speakers for $1. This is part 1 of the bug. Part 2 is that when I'm watching movies using the 5.1 Output, some of the movie sounds are missing because ALSA transfers them to the non-existing speakers. It's also interesting that with 5.1 Analog Output listening to music (Audacious) the sound is perfect - no missing parts and stuff, the problem happens only when I'm watching movies. I tried different media players but it's the same with all of them.* *If there's any additional information I could give you, such as hardware info, please let me know.*

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Re: [alsa-devel] [PATCH -next] platform/chrome: Fix Kconfig dependencies for wilco_ec
by Enric Balletbo i Serra 25 Feb '19

25 Feb '19

Hi Randy, Thanks for the report On 22/2/19 17:50, Randy Dunlap wrote: > On 2/20/19 11:57 PM, Enric Balletbo i Serra wrote: >> Hi, >> >> On 21/2/19 0:09, Randy Dunlap wrote: >>> On 2/20/19 2:11 PM, Nick Crews wrote: >>>> In the initial version of the Wilco EC Driver, the >>>> dependency order was wrong. It before was possible to >>>> select CONFIG_WILCO_EC and CONFIG_CROS_EC_LPC without >>>> having CONFIG_CROS_EC_LPC_MEC. This was wrong, since >>>> WILCO_EC depends upon CONFIG CROS_EC_LPC_MEC, not the >>>> other way around. >>>> >>>> Fixes: 1733c32834e5d1 ("platform/chrome: Add new driver for Wilco EC") >>>> Signed-off-by: Nick Crews <ncrews(a)chromium.org> >>> >>> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> >>> Acked-by: Randy Dunlap <rdunlap(a)infradead.org> # build-tested >>> >> >> As this is [-next] material I squashed that commit and queued for 5.1 > > Hi, > > on linux-next of 20190222, I now see this build error + Kconfig warnings: > > (this is on i386; randconfig file is attached) > > WARNING: unmet direct dependencies detected for SND_SOC_CROS_EC_CODEC > Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_CROS_EC [=n] > Selected by [y]: > - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] This error seems to be introduced by following commit. commit 74c6ecf4194ebed285b29964a950e0cd7414fe19 Author: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Mon Feb 18 12:18:19 2019 +0800 ASoC: qcom: Kconfig: select dmic for sdm845 sdm845 uses dmic on EC so it should select CROS_EC_CODEC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> The problem is that this commit selects CROS_EC_CODEC but the CROS_EC_CODEC depends on MFD_CROS_EC which is not selected. A question for Cheng-Yi, I suppose that the answer is yes but worth ask because I don't know about this device. Is the sdm845 a chromebook with and embedded controller right? Thanks, Enric > > ld: sound/soc/codecs/cros_ec_codec.o: in function `ec_command_no_resp': > cros_ec_codec.c:(.text+0x50): undefined reference to `cros_ec_cmd_xfer_status' > ld: sound/soc/codecs/cros_ec_codec.o: in function `mic_gain_get': > cros_ec_codec.c:(.text+0x5a8): undefined reference to `cros_ec_cmd_xfer_status' > > >

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[alsa-devel] [PATCH v4 00/14] ASoC: Sound Open Firmware (SOF) core
by Pierre-Louis Bossart 25 Feb '19

25 Feb '19

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core and utilities. Support for Intel devices is provided as a follow-up series. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Changes since v3: Addressed dozens ofcomments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Hardened memory allocation, fixed module load/unload oops or errors (now at hundreds of cycles on ApolloLake devices) Fixed suspend/resume and removed use of suspend_late, now using snd_soc_pm_ops and standard flow Fixed topology free (multiple patches already contributed for ASoC-core) Fixed debugfs/pm_runtime interaction Removed error checks on debugfs, changed to EXPORT_SYMBOL_GPL Added filenames in debug messages, ues hex_dump_to_buffer Removed "sof-audio" platform device and added better error checks on probe Reworked data structures to remove mix of const/variable fields and duplication of pointers for platform-specific changes Removed redundant ops tables Fixed trace/logger issues Fixed issues with loader (alignment and block size errors) Fixed release_firmware handling (was all over the place) Simplified PCI handling Added comments on non-atomic triggers Removed enum controls (not supported in topology) Simplified include file dependencies Fix cppcheck warnings Fixed Kconfigs to deal with Kbuild warnings on exotic architectures Precisions: The code in this patchset is directly squashed from the SOF development branch [1], which tracks Mark Brown's for-next branch on a weekly basis and the configurations used for testing are based on the defconfigs at [2]. The patches are also used backported for Chromebook devices. Full disclosure on known limitation and issues (full list at [3]) a) the interaction with ASoC is not always perfect, there are a couple of points where a better solution is desired (use of private data, support for link DMA, etc). The known points are explicitly documented in the code and will be updated. To be clearer, it's our belief that SOF does not cripple ASoC in any way, and that merging this SOF core does not harm others. It's just complicated to get things right, especially on a couple of cases where the topology and DPCM frameworks generate complex flows that very few people understand. b) The get/put methods for controls generate an IPC and possibly a wake-up. This is not optimal and is being fixed. c) runtime_pm is being hardened and the use of SMART_SUSPEND was suggested. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. This patchset also includes contributions from Daniel Baluta for loading code on non-Intel platforms. Pierre [1] https://github.com/thesofproject/linux [2] https://github.com/thesofproject/kconfig [3] https://github.com/thesofproject/linux/issues Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware logger support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add PM support ASoC: SOF: Add Nocodec machine driver support Pierre-Louis Bossart (2): ASoC: SOF: Add xtensa support ASoC: SOF: Add utils include/sound/soc.h | 3 + include/sound/sof.h | 100 + include/sound/sof/control.h | 125 ++ include/sound/sof/dai-intel.h | 178 ++ include/sound/sof/dai.h | 75 + include/sound/sof/header.h | 158 ++ include/sound/sof/info.h | 118 ++ include/sound/sof/pm.h | 48 + include/sound/sof/stream.h | 149 ++ include/sound/sof/topology.h | 256 +++ include/sound/sof/trace.h | 66 + include/sound/sof/xtensa.h | 44 + include/uapi/sound/sof/abi.h | 62 + include/uapi/sound/sof/eq.h | 164 ++ include/uapi/sound/sof/fw.h | 78 + include/uapi/sound/sof/header.h | 27 + include/uapi/sound/sof/manifest.h | 188 ++ include/uapi/sound/sof/tokens.h | 98 + include/uapi/sound/sof/tone.h | 21 + include/uapi/sound/sof/trace.h | 93 + sound/soc/sof/control.c | 361 ++++ sound/soc/sof/core.c | 473 +++++ sound/soc/sof/debug.c | 219 +++ sound/soc/sof/ipc.c | 842 +++++++++ sound/soc/sof/loader.c | 363 ++++ sound/soc/sof/nocodec.c | 109 ++ sound/soc/sof/ops.c | 204 +++ sound/soc/sof/ops.h | 414 +++++ sound/soc/sof/pcm.c | 710 ++++++++ sound/soc/sof/pm.c | 381 ++++ sound/soc/sof/sof-priv.h | 599 ++++++ sound/soc/sof/topology.c | 2808 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 298 +++ sound/soc/sof/utils.c | 109 ++ sound/soc/sof/xtensa/Kconfig | 2 + sound/soc/sof/xtensa/Makefile | 5 + sound/soc/sof/xtensa/core.c | 138 ++ 37 files changed, 10086 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/topology.h create mode 100644 include/sound/sof/trace.h create mode 100644 include/sound/sof/xtensa.h create mode 100644 include/uapi/sound/sof/abi.h create mode 100644 include/uapi/sound/sof/eq.h create mode 100644 include/uapi/sound/sof/fw.h create mode 100644 include/uapi/sound/sof/header.h create mode 100644 include/uapi/sound/sof/manifest.h create mode 100644 include/uapi/sound/sof/tokens.h create mode 100644 include/uapi/sound/sof/tone.h create mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/nocodec.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c create mode 100644 sound/soc/sof/utils.c create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c -- 2.17.1

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[alsa-devel] [PATCH] arm64: dts: renesas: r8a7796: remove unneeded sound #address/size-cells
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> commit 78bc93b3ffb2 ("arm64: dts: renesas: r8a7796: Add address properties to rcar_sound port nodes") added missing #address-cells and #size-cells for sound ports. But, these are based on platform, not on SoC. This patch cleanups it. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts | 1 + arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts | 1 + arch/arm64/boot/dts/renesas/r8a7796.dtsi | 11 ----------- 3 files changed, 2 insertions(+), 11 deletions(-) diff --git a/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts b/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts index b4f9567..2aefa53 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts +++ b/arch/arm64/boot/dts/renesas/r8a7796-salvator-x.dts @@ -68,6 +68,7 @@ ports { /* rsnd_port0 is on salvator-common */ rsnd_port1: port@1 { + reg = <1>; rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; diff --git a/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts b/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts index 8860be6..fa4ef7c 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts +++ b/arch/arm64/boot/dts/renesas/r8a7796-salvator-xs.dts @@ -69,6 +69,7 @@ ports { /* rsnd_port0 is on salvator-common */ rsnd_port1: port@1 { + reg = <1>; rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; diff --git a/arch/arm64/boot/dts/renesas/r8a7796.dtsi b/arch/arm64/boot/dts/renesas/r8a7796.dtsi index 0648d12..cc601d9 100644 --- a/arch/arm64/boot/dts/renesas/r8a7796.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a7796.dtsi @@ -2162,17 +2162,6 @@ dma-names = "rx", "tx", "rxu", "txu"; }; }; - - ports { - #address-cells = <1>; - #size-cells = <0>; - port@0 { - reg = <0>; - }; - port@1 { - reg = <1>; - }; - }; }; audma0: dma-controller@ec700000 { -- 2.7.4

3 4

[alsa-devel] [PATCH] arm64: dts: renesas: r8a77965: add SSIU support for sound
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> rsnd driver supports SSIU now, let's use it. Then, BUSIF DMA settings on rcar_sound,ssi (= rxu, txu) are no longer needed. Applies commit 8d14bfa074db ("arm64: dts: renesas: r8a7796: add SSIU support for sound") for r8a77965. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- arch/arm64/boot/dts/renesas/r8a77965.dtsi | 211 ++++++++++++++++++++++++++++++ 1 file changed, 211 insertions(+) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index 4b3730f..cc77c2e 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1585,6 +1585,217 @@ }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + ssiu02: ssiu-2 { + dmas = <&audma0 0x37>, <&audma1 0x38>; + dma-names = "rx", "tx"; + }; + ssiu03: ssiu-3 { + dmas = <&audma0 0x47>, <&audma1 0x48>; + dma-names = "rx", "tx"; + }; + ssiu04: ssiu-4 { + dmas = <&audma0 0x3F>, <&audma1 0x40>; + dma-names = "rx", "tx"; + }; + ssiu05: ssiu-5 { + dmas = <&audma0 0x43>, <&audma1 0x44>; + dma-names = "rx", "tx"; + }; + ssiu06: ssiu-6 { + dmas = <&audma0 0x4F>, <&audma1 0x50>; + dma-names = "rx", "tx"; + }; + ssiu07: ssiu-7 { + dmas = <&audma0 0x53>, <&audma1 0x54>; + dma-names = "rx", "tx"; + }; + ssiu10: ssiu-8 { + dmas = <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx"; + }; + ssiu11: ssiu-9 { + dmas = <&audma0 0x4B>, <&audma1 0x4C>; + dma-names = "rx", "tx"; + }; + ssiu12: ssiu-10 { + dmas = <&audma0 0x57>, <&audma1 0x58>; + dma-names = "rx", "tx"; + }; + ssiu13: ssiu-11 { + dmas = <&audma0 0x59>, <&audma1 0x5A>; + dma-names = "rx", "tx"; + }; + ssiu14: ssiu-12 { + dmas = <&audma0 0x5F>, <&audma1 0x60>; + dma-names = "rx", "tx"; + }; + ssiu15: ssiu-13 { + dmas = <&audma0 0xC3>, <&audma1 0xC4>; + dma-names = "rx", "tx"; + }; + ssiu16: ssiu-14 { + dmas = <&audma0 0xC7>, <&audma1 0xC8>; + dma-names = "rx", "tx"; + }; + ssiu17: ssiu-15 { + dmas = <&audma0 0xCB>, <&audma1 0xCC>; + dma-names = "rx", "tx"; + }; + ssiu20: ssiu-16 { + dmas = <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx"; + }; + ssiu21: ssiu-17 { + dmas = <&audma0 0x67>, <&audma1 0x68>; + dma-names = "rx", "tx"; + }; + ssiu22: ssiu-18 { + dmas = <&audma0 0x6B>, <&audma1 0x6C>; + dma-names = "rx", "tx"; + }; + ssiu23: ssiu-19 { + dmas = <&audma0 0x6D>, <&audma1 0x6E>; + dma-names = "rx", "tx"; + }; + ssiu24: ssiu-20 { + dmas = <&audma0 0xCF>, <&audma1 0xCE>; + dma-names = "rx", "tx"; + }; + ssiu25: ssiu-21 { + dmas = <&audma0 0xEB>, <&audma1 0xEC>; + dma-names = "rx", "tx"; + }; + ssiu26: ssiu-22 { + dmas = <&audma0 0xED>, <&audma1 0xEE>; + dma-names = "rx", "tx"; + }; + ssiu27: ssiu-23 { + dmas = <&audma0 0xEF>, <&audma1 0xF0>; + dma-names = "rx", "tx"; + }; + ssiu30: ssiu-24 { + dmas = <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx"; + }; + ssiu31: ssiu-25 { + dmas = <&audma0 0x21>, <&audma1 0x22>; + dma-names = "rx", "tx"; + }; + ssiu32: ssiu-26 { + dmas = <&audma0 0x23>, <&audma1 0x24>; + dma-names = "rx", "tx"; + }; + ssiu33: ssiu-27 { + dmas = <&audma0 0x25>, <&audma1 0x26>; + dma-names = "rx", "tx"; + }; + ssiu34: ssiu-28 { + dmas = <&audma0 0x27>, <&audma1 0x28>; + dma-names = "rx", "tx"; + }; + ssiu35: ssiu-29 { + dmas = <&audma0 0x29>, <&audma1 0x2A>; + dma-names = "rx", "tx"; + }; + ssiu36: ssiu-30 { + dmas = <&audma0 0x2B>, <&audma1 0x2C>; + dma-names = "rx", "tx"; + }; + ssiu37: ssiu-31 { + dmas = <&audma0 0x2D>, <&audma1 0x2E>; + dma-names = "rx", "tx"; + }; + ssiu40: ssiu-32 { + dmas = <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx"; + }; + ssiu41: ssiu-33 { + dmas = <&audma0 0x17>, <&audma1 0x18>; + dma-names = "rx", "tx"; + }; + ssiu42: ssiu-34 { + dmas = <&audma0 0x19>, <&audma1 0x1A>; + dma-names = "rx", "tx"; + }; + ssiu43: ssiu-35 { + dmas = <&audma0 0x1B>, <&audma1 0x1C>; + dma-names = "rx", "tx"; + }; + ssiu44: ssiu-36 { + dmas = <&audma0 0x1D>, <&audma1 0x1E>; + dma-names = "rx", "tx"; + }; + ssiu45: ssiu-37 { + dmas = <&audma0 0x1F>, <&audma1 0x20>; + dma-names = "rx", "tx"; + }; + ssiu46: ssiu-38 { + dmas = <&audma0 0x31>, <&audma1 0x32>; + dma-names = "rx", "tx"; + }; + ssiu47: ssiu-39 { + dmas = <&audma0 0x33>, <&audma1 0x34>; + dma-names = "rx", "tx"; + }; + ssiu50: ssiu-40 { + dmas = <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx"; + }; + ssiu60: ssiu-41 { + dmas = <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx"; + }; + ssiu70: ssiu-42 { + dmas = <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx"; + }; + ssiu80: ssiu-43 { + dmas = <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx"; + }; + ssiu90: ssiu-44 { + dmas = <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx"; + }; + ssiu91: ssiu-45 { + dmas = <&audma0 0x7F>, <&audma1 0x80>; + dma-names = "rx", "tx"; + }; + ssiu92: ssiu-46 { + dmas = <&audma0 0x81>, <&audma1 0x82>; + dma-names = "rx", "tx"; + }; + ssiu93: ssiu-47 { + dmas = <&audma0 0x83>, <&audma1 0x84>; + dma-names = "rx", "tx"; + }; + ssiu94: ssiu-48 { + dmas = <&audma0 0xA3>, <&audma1 0xA4>; + dma-names = "rx", "tx"; + }; + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 25 Feb '19

25 Feb '19

From: Shengjiu Wang <shengjiu.wang(a)freescale.com> 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 begginning with the normal enable flow. This is hardware issue, 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 Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..23bd0ad4ac31 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: @@ -611,12 +606,23 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, 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 +912,12 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + /* 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 2

[alsa-devel] simple-audio-card,mclk-fs
by Daniel Baluta 25 Feb '19

25 Feb '19

Hello Morimoto-san, Do you have any plans on adding support for multiple mclk-fs values in simple-card? Usually one constant mclk-fs value is not enough because usually codecs have different value depending on rate. For example, for wm8524 [1] one can use mclk-fs = 256 for rates between 8kHz and 96kHz but it needs to change mclk-fs = 128 for 176.4hHz and 192kHz. I am thinking of something like this: --- a/arch/arm64/boot/dts/freescale/imx8mq-evk.dts +++ b/arch/arm64/boot/dts/freescale/imx8mq-evk.dts @@ -44,6 +44,8 @@ compatible = "simple-audio-card"; simple-audio-card,name = "wm8524-audio"; simple-audio-card,format = "i2s"; + simple-audio-card,lrclk-fs = <8000>, <32000>, <44100> <48000>, <96000>, <176400>, <192000>; + simple-audio-card,mclk-fs = <256>, <256>, <256>, <256>, <256>, <128>, <128>; Let me know if this would be a good direction and I will send a patch. thanks, Daniel. [1] https://www.cirrus.com/products/wm8524/

2 1

[alsa-devel] [PATCH] ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
by twischer＠de.adit-jv.com 25 Feb '19

25 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif mode/adinr/dalign address is registered by: (in busif4 case) RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80) But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif mode/adinr/dalign register address ( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] ) are out of this rule. This patch registers ssi9 4/5/6/7 mode/adinr/dalign register as single register, and access these these registers in case of SSI9 BUSIF 4/5/6/7. Fixes: commit 8c9d75033340 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/gen.c | 24 ++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 27 +++++++++++++++++++++++++++ sound/soc/sh/rcar/ssiu.c | 36 ++++++++++++++++-------------------- 3 files changed, 67 insertions(+), 20 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 7cda601..af19010 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -255,6 +255,30 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c), + RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484), + RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488), + RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0), + RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4), + RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8), + RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0), + RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4), + RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8), + RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0), + RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4), + RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8), + RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80), + RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84), + RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88), + RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0), + RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4), + RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8), + RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0), + RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4), + RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8), + RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0), + RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4), + RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8), }; static const struct rsnd_regmap_field_conf conf_scu[] = { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 605e4b9..90625c5 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -191,6 +191,30 @@ enum rsnd_reg { SSI_SYS_STATUS7, HDMI0_SEL, HDMI1_SEL, + SSI9_BUSIF0_MODE, + SSI9_BUSIF1_MODE, + SSI9_BUSIF2_MODE, + SSI9_BUSIF3_MODE, + SSI9_BUSIF4_MODE, + SSI9_BUSIF5_MODE, + SSI9_BUSIF6_MODE, + SSI9_BUSIF7_MODE, + SSI9_BUSIF0_ADINR, + SSI9_BUSIF1_ADINR, + SSI9_BUSIF2_ADINR, + SSI9_BUSIF3_ADINR, + SSI9_BUSIF4_ADINR, + SSI9_BUSIF5_ADINR, + SSI9_BUSIF6_ADINR, + SSI9_BUSIF7_ADINR, + SSI9_BUSIF0_DALIGN, + SSI9_BUSIF1_DALIGN, + SSI9_BUSIF2_DALIGN, + SSI9_BUSIF3_DALIGN, + SSI9_BUSIF4_DALIGN, + SSI9_BUSIF5_DALIGN, + SSI9_BUSIF6_DALIGN, + SSI9_BUSIF7_DALIGN, /* SSI */ SSICR, @@ -209,6 +233,9 @@ enum rsnd_reg { #define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) #define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) #define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI9_BUSIF_MODE(i) (SSI9_BUSIF0_MODE + (i)) +#define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) +#define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index c74991d..c036654 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -181,29 +181,25 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); int busif = rsnd_mod_id_sub(mod); + int chnl = rsnd_io_is_play(io) ? + rsnd_runtime_channel_after_ctu(io) : + rsnd_runtime_channel_original(io); - /* - * FIXME - * - * We can't support SSI9-4/5/6/7, because its address is - * out of calculation rule - */ if ((id == 9) && (busif >= 4)) { - struct device *dev = rsnd_priv_to_dev(priv); - - dev_err(dev, "This driver doesn't support SSI%d-%d, so far", - id, busif); + rsnd_mod_write(mod, SSI9_BUSIF_ADINR(busif), + rsnd_get_adinr_bit(mod, io) | chnl); + rsnd_mod_write(mod, SSI9_BUSIF_MODE(busif), + rsnd_get_busif_shift(io, mod) | 1); + rsnd_mod_write(mod, SSI9_BUSIF_DALIGN(busif), + rsnd_get_dalign(mod, io)); + } else { + rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), + rsnd_get_adinr_bit(mod, io) | chnl); + rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), + rsnd_get_busif_shift(io, mod) | 1); + rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), + rsnd_get_dalign(mod, io)); } - - rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), - rsnd_get_adinr_bit(mod, io) | - (rsnd_io_is_play(io) ? - rsnd_runtime_channel_after_ctu(io) : - rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), - rsnd_get_busif_shift(io, mod) | 1); - rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), - rsnd_get_dalign(mod, io)); } if (has_hdmi0 || has_hdmi1) { -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: remove unsupported formats in i2s master mode
by Sven Van Asbroeck 23 Feb '19

23 Feb '19

Quote from the ssi manual: "The word length is fixed to 32 in I2S Master mode and the WL bits determine the number of bits that will contain valid data (out of the 32 transmitted/received bits in each channel)." (61.8.1.4, page 5138, IMX6DQRM Rev. 3, 07/2015) In other words, in i2s master mode, we always get 32 physical bits/channel, no matter what. However, the supported list of formats for this ssi contains S8, S16_XX, S18_3XX, and S20_3XX, which have 8, 16, 24 and 24 physical bits/channel respectively. If one of these is selected in i2s master mode, the format generated by the ssi will be incorrect. Fix by constraining the frame bits in i2s master mode to 32 * channels. This will filter out those formats that the ssi cannot support. Tested on imx6 with tda998x and sgtl5k codecs. 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> Signed-off-by: Sven Van Asbroeck <TheSven73(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0a648229e643..3d375f60ecf4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -627,6 +627,28 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi *ssi) regmap_write(regs, REG_SSI_SOR, SSI_SOR_WAIT(3)); } +static int fsl_ssi_hw_rule_i2s_master(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *it = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_FRAME_BITS); + struct fsl_ssi *ssi = rule->private; + struct snd_interval t; + + if (!fsl_ssi_is_i2s_master(ssi)) + return 0; + + /* + * In i2s master mode, the ssi always generates 32 physical + * bits/channel. Filter out formats that don't have 32 physical + * bits/channel, they are unsupported. + */ + memset(&t, 0, sizeof(t)); + t.min = t.max = params_channels(params) * 32; + + return snd_interval_refine(it, &t); +} + static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -648,6 +670,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2); + snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + fsl_ssi_hw_rule_i2s_master, ssi, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); + return 0; } -- 2.17.1

1 1

[alsa-devel] [PATCH 0/2] ASoC: adau1977: Add support for micbias in DT
by Bogdan Togorean 22 Feb '19

22 Feb '19

Currently micbias voltage is set based on platform_data. If driver is instantiated using DT only default voltage for micbias is possible. This patch adds the possibility to specify an optional DT property to configure the output voltage for micbias. Bogdan Togorean (2): ASoC: adau1977: Add MICBIAS example in DT bindings ASoC: adau1977: Add support for setting MICBIAS via DT .../devicetree/bindings/sound/adi,adau1977.txt | 7 +++++++ sound/soc/codecs/adau1977.c | 12 +++++++----- 2 files changed, 14 insertions(+), 5 deletions(-) -- 2.20.1

3 5

[alsa-devel] [PATCH] printk: Remove no longer used LOG_PREFIX.
by Tetsuo Handa 22 Feb '19

22 Feb '19

When commit 5becfb1df5ac8e49 ("kmsg: merge continuation records while printing") introduced LOG_PREFIX, we used KERN_DEFAULT etc. as a flag for setting LOG_PREFIX in order to tell whether to call cont_add() (i.e. whether to append the message to "struct cont"). But since commit 4bcc595ccd80decb ("printk: reinstate KERN_CONT for printing continuation lines") inverted the behavior (i.e. don't append the message to "struct cont" unless KERN_CONT is specified) and commit 5aa068ea4082b39e ("printk: remove games with previous record flags") removed the last LOG_PREFIX check, setting LOG_PREFIX via KERN_DEFAULT etc. is no longer meaningful. Therefore, we can remove LOG_PREFIX and make KERN_DEFAULT empty string. Signed-off-by: Tetsuo Handa <penguin-kernel(a)I-love.SAKURA.ne.jp> --- include/linux/kern_levels.h | 2 +- include/linux/printk.h | 1 - kernel/printk/printk.c | 6 +----- 3 files changed, 2 insertions(+), 7 deletions(-) diff --git a/include/linux/kern_levels.h b/include/linux/kern_levels.h index d237fe8..bf2389c 100644 --- a/include/linux/kern_levels.h +++ b/include/linux/kern_levels.h @@ -14,7 +14,7 @@ #define KERN_INFO KERN_SOH "6" /* informational */ #define KERN_DEBUG KERN_SOH "7" /* debug-level messages */ -#define KERN_DEFAULT KERN_SOH "d" /* the default kernel loglevel */ +#define KERN_DEFAULT "" /* the default kernel loglevel */ /* * Annotation for a "continued" line of log printout (only done after a diff --git a/include/linux/printk.h b/include/linux/printk.h index 77740a5..9fe6c9e 100644 --- a/include/linux/printk.h +++ b/include/linux/printk.h @@ -18,7 +18,6 @@ static inline int printk_get_level(const char *buffer) if (buffer[0] == KERN_SOH_ASCII && buffer[1]) { switch (buffer[1]) { case '0' ... '7': - case 'd': /* KERN_DEFAULT */ case 'c': /* KERN_CONT */ return buffer[1]; } diff --git a/kernel/printk/printk.c b/kernel/printk/printk.c index d3d1703..3e4a735 100644 --- a/kernel/printk/printk.c +++ b/kernel/printk/printk.c @@ -344,7 +344,6 @@ enum con_msg_format_flags { enum log_flags { LOG_NEWLINE = 2, /* text ended with a newline */ - LOG_PREFIX = 4, /* text started with a prefix */ LOG_CONT = 8, /* text is a fragment of a continuation line */ }; @@ -1867,9 +1866,6 @@ int vprintk_store(int facility, int level, case '0' ... '7': if (level == LOGLEVEL_DEFAULT) level = kern_level - '0'; - /* fallthrough */ - case 'd': /* KERN_DEFAULT */ - lflags |= LOG_PREFIX; break; case 'c': /* KERN_CONT */ lflags |= LOG_CONT; @@ -1884,7 +1880,7 @@ int vprintk_store(int facility, int level, level = default_message_loglevel; if (dict) - lflags |= LOG_PREFIX|LOG_NEWLINE; + lflags |= LOG_NEWLINE; return log_output(facility, level, lflags, dict, dictlen, text, text_len); -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda/tegra: property for card name
by Sameer Pujar 22 Feb '19

22 Feb '19

An optional property "nvidia,model" is introduced for hda to pass custom name for the sound card. The suffix "-hda" in the name passed is useful to distinguish between multiple cards available for a platform. When the property is not specified, default name("tegra-hda") mentioned in hda driver is used. This property can be added in platform specific file of the board and card name can relate to the board in use. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Jonathan Hunter <jonathanh(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt index 44d2745..21cd310 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -13,6 +13,10 @@ Required properties: See ../reset/reset.txt for details. - reset-names : Must include the following entries: hda, hda2hdmi, hda2codec_2x +Optional properties: +- nvidia,model : The user-visible name of this sound complex. Since the property + is optional, legacy boards can use default name provided in hda driver. + Example: hda@70030000 { @@ -27,4 +31,5 @@ hda@70030000 { <&tegra_car 128>, /* hda2hdmi */ <&tegra_car 111>; /* hda2codec_2x */ reset-names = "hda", "hda2hdmi", "hda2codec_2x"; + nvidia,model = "jetson-tk1-hda"; }; -- 2.7.4

2 5

[alsa-devel] doc pages don't work
by Jarek 22 Feb '19

22 Feb '19

Hi, https://www.alsa-project.org/alsa-doc/alsa-lib/ return 404 Jarek

2 1

[alsa-devel] Playback samples at a specific time on Intel HDA
by Rajagopal Aravindan 21 Feb '19

21 Feb '19

Hello, I am trying to accomplish synchronized audio playback using ALSA & gPTP, on an Intel SkyLake platform have Intel HDA. I do understand that, starting with SkyLake platform, there is support for querying link time. But, how do I instruct to play the samples at a given (gPTP) time ? Is this possible ? Can someone please point me to code or doc, relevant to my goal ? In essence, I would like to play the samples at the stroke of a given (gPTP) time. Also, the ALSA page says ... "The link time can be directly measured if supported in hardware by sample counters or wallclocks (e.g. with HDAudio 24MHz or PTP clock for networked solutions)" What is the connection between PTP clock and sound card here, are there sound cards that can latch on to a PTP clock on a NIC or what ? Thanks in advance. Regards, Rajagopal

2 1

[alsa-devel] Regarding alsa-utils commit "alsaucm: add alsa-ucm udev rules for PAZ00 (Toshiba AC100/Dynabook AZ)."
by Tanu Kaskinen 21 Feb '19

21 Feb '19

Hi Jaroslav, I'm updating alsa-utils in OpenEmbedded, and I noticed that there's a new udev rules file, 89-alsa-ucm.rules. It seems to just initialize the mixer settings for particular hardware (Toshiba AC100/Dynabook AZ), and doing that in alsa-utils seems weird. Why does this hardware use a udev rules file for initialization, unlike any other hardware? I think it would be good to have an explanatory comment in the rules file. -- Tanu https://www.patreon.com/tanuk https://liberapay.com/tanuk

2 2

[alsa-devel] [RFC 0/4] clk/driver: platform: Fix kfree() of const memory on setting driver_override
by Krzysztof Kozlowski 21 Feb '19

21 Feb '19

Hi, The problem =========== Several device types (platform, amba, spi etc.) provide a driver_override field. On sysfs store or during device removal, they kfree() the existing value. However the users are unaware of this and set the driver_override like: pdev->driver_override = "exynos5-subcmu"; which obviously leads to error. Solution ======== I provided simple helper for platform device. If this approach is acceptable, I can convert also other buses, like AMBA, SPI. Dependencies and pick up order ============================== Patch 1: please pick it as is through clock tree Patch 3: Depends on patch 1 (merge conflict) and 2. Patch 4: Depends on patch 2. Best regards, Krzysztof Krzysztof Kozlowski (4): clk: samsung: exynos5: Fix possible NULL pointer exception on platform_device_alloc() failure driver: platform: Provide helper for safer setting of driver_override clk: samsung: exynos5: Fix kfree() of const memory on setting driver_override slimbus: ngd: Fix kfree() of const memory on setting driver_override drivers/base/platform.c | 63 ++++++++++++++++++++++---------- drivers/clk/samsung/clk-exynos5-subcmu.c | 12 ++++-- drivers/slimbus/qcom-ngd-ctrl.c | 2 +- include/linux/platform_device.h | 9 ++++- 4 files changed, 61 insertions(+), 25 deletions(-) -- 2.7.4

3 8

[alsa-devel] [PATCH] ASoC: samsung: odroid: Prevent uninitialized variable use
by Sylwester Nawrocki 21 Feb '19

21 Feb '19

This addresses an issue pointed out by compiler warning: sound/soc/samsung/odroid.c: In function ‘odroid_audio_probe’: sound/soc/samsung/odroid.c:298:22: warning: ‘cpu_dai’ may be used uninitialized in this function [-Wmaybe-uninitialized] priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis"); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index c3b0f6c612cb..694512f980fd 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -194,7 +194,8 @@ static struct snd_soc_dai_link odroid_card_dais[] = { static int odroid_audio_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct device_node *cpu, *cpu_dai, *codec; + struct device_node *cpu_dai = NULL; + struct device_node *cpu, *codec; struct odroid_priv *priv; struct snd_soc_card *card; struct snd_soc_dai_link *link, *codec_link; @@ -271,8 +272,11 @@ static int odroid_audio_probe(struct platform_device *pdev) if (ret < 0) break; } - if (ret == 0) + if (ret == 0) { cpu_dai = of_parse_phandle(cpu, "sound-dai", 0); + if (!cpu_dai) + ret = -EINVAL; + } of_node_put(cpu); of_node_put(codec); -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX362FA with ALC294
by Jian-Hong Pan 21 Feb '19

21 Feb '19

The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs through the internal speaker and the headphone. This issue can be fixed by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294. Besides, ASUS UX362FA and UX533FD have the same audio initial pin config values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Ming Shuo Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ffa36e987b4..6421864a8a0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6771,7 +6771,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), - SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), @@ -7388,6 +7387,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_SPK, + {0x12, 0x90a60130}, + {0x17, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_SPK, {0x12, 0x90a60130}, {0x17, 0x90170110}, -- 2.20.1

2 1

[alsa-devel] [asoc:for-5.1 255/257] sound/soc/samsung/odroid.c:298:22: warning: 'cpu_dai' may be used uninitialized in this function
by kbuild test robot 20 Feb '19

20 Feb '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.1 head: 70b773219a32c7b8f3e53e041bc023ad99fd81f4 commit: d832d2b246c516eacb2d0ba53ec17ed59c3cd62b [255/257] ASoC: samsung: odroid: Fix of_node refcount unbalance config: sh-allmodconfig (attached as .config) compiler: sh4-linux-gnu-gcc (Debian 8.2.0-11) 8.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout d832d2b246c516eacb2d0ba53ec17ed59c3cd62b # save the attached .config to linux build tree GCC_VERSION=8.2.0 make.cross ARCH=sh Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/samsung/odroid.c: In function 'odroid_audio_probe': >> sound/soc/samsung/odroid.c:298:22: warning: 'cpu_dai' may be used uninitialized in this function [-Wmaybe-uninitialized] priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis"); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/cpu_dai +298 sound/soc/samsung/odroid.c aba611fc Sylwester Nawrocki 2017-04-21 193 aba611fc Sylwester Nawrocki 2017-04-21 194 static int odroid_audio_probe(struct platform_device *pdev) aba611fc Sylwester Nawrocki 2017-04-21 195 { aba611fc Sylwester Nawrocki 2017-04-21 196 struct device *dev = &pdev->dev; bc3cf17b Sylwester Nawrocki 2019-02-14 197 struct device_node *cpu, *cpu_dai, *codec; aba611fc Sylwester Nawrocki 2017-04-21 198 struct odroid_priv *priv; aba611fc Sylwester Nawrocki 2017-04-21 199 struct snd_soc_card *card; bc3cf17b Sylwester Nawrocki 2019-02-14 200 struct snd_soc_dai_link *link, *codec_link; bc3cf17b Sylwester Nawrocki 2019-02-14 201 int num_pcms, ret, i; bc3cf17b Sylwester Nawrocki 2019-02-14 202 struct of_phandle_args args = {}; aba611fc Sylwester Nawrocki 2017-04-21 203 aba611fc Sylwester Nawrocki 2017-04-21 204 priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); aba611fc Sylwester Nawrocki 2017-04-21 205 if (!priv) aba611fc Sylwester Nawrocki 2017-04-21 206 return -ENOMEM; aba611fc Sylwester Nawrocki 2017-04-21 207 aba611fc Sylwester Nawrocki 2017-04-21 208 card = &priv->card; aba611fc Sylwester Nawrocki 2017-04-21 209 card->dev = dev; aba611fc Sylwester Nawrocki 2017-04-21 210 aba611fc Sylwester Nawrocki 2017-04-21 211 card->owner = THIS_MODULE; aba611fc Sylwester Nawrocki 2017-04-21 212 card->fully_routed = true; aba611fc Sylwester Nawrocki 2017-04-21 213 b5c16a24 Sylwester Nawrocki 2019-02-14 214 spin_lock_init(&priv->lock); aba611fc Sylwester Nawrocki 2017-04-21 215 snd_soc_card_set_drvdata(card, priv); aba611fc Sylwester Nawrocki 2017-04-21 216 aba611fc Sylwester Nawrocki 2017-04-21 217 ret = snd_soc_of_parse_card_name(card, "model"); aba611fc Sylwester Nawrocki 2017-04-21 218 if (ret < 0) aba611fc Sylwester Nawrocki 2017-04-21 219 return ret; aba611fc Sylwester Nawrocki 2017-04-21 220 aba611fc Sylwester Nawrocki 2017-04-21 221 if (of_property_read_bool(dev->of_node, "samsung,audio-widgets")) { aba611fc Sylwester Nawrocki 2017-04-21 222 ret = snd_soc_of_parse_audio_simple_widgets(card, aba611fc Sylwester Nawrocki 2017-04-21 223 "samsung,audio-widgets"); aba611fc Sylwester Nawrocki 2017-04-21 224 if (ret < 0) aba611fc Sylwester Nawrocki 2017-04-21 225 return ret; aba611fc Sylwester Nawrocki 2017-04-21 226 } aba611fc Sylwester Nawrocki 2017-04-21 227 aba611fc Sylwester Nawrocki 2017-04-21 228 if (of_property_read_bool(dev->of_node, "samsung,audio-routing")) { aba611fc Sylwester Nawrocki 2017-04-21 229 ret = snd_soc_of_parse_audio_routing(card, aba611fc Sylwester Nawrocki 2017-04-21 230 "samsung,audio-routing"); aba611fc Sylwester Nawrocki 2017-04-21 231 if (ret < 0) aba611fc Sylwester Nawrocki 2017-04-21 232 return ret; aba611fc Sylwester Nawrocki 2017-04-21 233 } aba611fc Sylwester Nawrocki 2017-04-21 234 bc3cf17b Sylwester Nawrocki 2019-02-14 235 card->dai_link = odroid_card_dais; bc3cf17b Sylwester Nawrocki 2019-02-14 236 card->num_links = ARRAY_SIZE(odroid_card_dais); aba611fc Sylwester Nawrocki 2017-04-21 237 aba611fc Sylwester Nawrocki 2017-04-21 238 cpu = of_get_child_by_name(dev->of_node, "cpu"); aba611fc Sylwester Nawrocki 2017-04-21 239 codec = of_get_child_by_name(dev->of_node, "codec"); bc3cf17b Sylwester Nawrocki 2019-02-14 240 link = card->dai_link; bc3cf17b Sylwester Nawrocki 2019-02-14 241 codec_link = &card->dai_link[1]; bc3cf17b Sylwester Nawrocki 2019-02-14 242 bc3cf17b Sylwester Nawrocki 2019-02-14 243 /* bc3cf17b Sylwester Nawrocki 2019-02-14 244 * For backwards compatibility create the secondary CPU DAI link only bc3cf17b Sylwester Nawrocki 2019-02-14 245 * if there are 2 CPU DAI entries in the cpu sound-dai property in DT. f89aea0f Sylwester Nawrocki 2019-02-15 246 * Also add required DAPM routes not available in old DTS. bc3cf17b Sylwester Nawrocki 2019-02-14 247 */ bc3cf17b Sylwester Nawrocki 2019-02-14 248 num_pcms = of_count_phandle_with_args(cpu, "sound-dai", bc3cf17b Sylwester Nawrocki 2019-02-14 249 "#sound-dai-cells"); f89aea0f Sylwester Nawrocki 2019-02-15 250 if (num_pcms == 1) { f89aea0f Sylwester Nawrocki 2019-02-15 251 card->dapm_routes = odroid_dapm_routes; f89aea0f Sylwester Nawrocki 2019-02-15 252 card->num_dapm_routes = ARRAY_SIZE(odroid_dapm_routes); bc3cf17b Sylwester Nawrocki 2019-02-14 253 card->num_links--; f89aea0f Sylwester Nawrocki 2019-02-15 254 } bc3cf17b Sylwester Nawrocki 2019-02-14 255 bc3cf17b Sylwester Nawrocki 2019-02-14 256 for (i = 0; i < num_pcms; i++, link += 2) { bc3cf17b Sylwester Nawrocki 2019-02-14 257 ret = of_parse_phandle_with_args(cpu, "sound-dai", bc3cf17b Sylwester Nawrocki 2019-02-14 258 "#sound-dai-cells", i, &args); bc3cf17b Sylwester Nawrocki 2019-02-14 259 if (ret < 0) d832d2b2 Sylwester Nawrocki 2019-02-20 260 break; aba611fc Sylwester Nawrocki 2017-04-21 261 bc3cf17b Sylwester Nawrocki 2019-02-14 262 if (!args.np) { bc3cf17b Sylwester Nawrocki 2019-02-14 263 dev_err(dev, "sound-dai property parse error: %d\n", ret); d832d2b2 Sylwester Nawrocki 2019-02-20 264 ret = -EINVAL; d832d2b2 Sylwester Nawrocki 2019-02-20 265 break; aba611fc Sylwester Nawrocki 2017-04-21 266 } aba611fc Sylwester Nawrocki 2017-04-21 267 bc3cf17b Sylwester Nawrocki 2019-02-14 268 ret = snd_soc_get_dai_name(&args, &link->cpu_dai_name); bc3cf17b Sylwester Nawrocki 2019-02-14 269 of_node_put(args.np); bc3cf17b Sylwester Nawrocki 2019-02-14 270 aba611fc Sylwester Nawrocki 2017-04-21 271 if (ret < 0) d832d2b2 Sylwester Nawrocki 2019-02-20 272 break; bc3cf17b Sylwester Nawrocki 2019-02-14 273 } d832d2b2 Sylwester Nawrocki 2019-02-20 274 if (ret == 0) bc3cf17b Sylwester Nawrocki 2019-02-14 275 cpu_dai = of_parse_phandle(cpu, "sound-dai", 0); d832d2b2 Sylwester Nawrocki 2019-02-20 276 bc3cf17b Sylwester Nawrocki 2019-02-14 277 of_node_put(cpu); bc3cf17b Sylwester Nawrocki 2019-02-14 278 of_node_put(codec); d832d2b2 Sylwester Nawrocki 2019-02-20 279 if (ret < 0) d832d2b2 Sylwester Nawrocki 2019-02-20 280 return ret; aba611fc Sylwester Nawrocki 2017-04-21 281 bc3cf17b Sylwester Nawrocki 2019-02-14 282 ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link); bc3cf17b Sylwester Nawrocki 2019-02-14 283 if (ret < 0) d832d2b2 Sylwester Nawrocki 2019-02-20 284 goto err_put_cpu_dai; aba611fc Sylwester Nawrocki 2017-04-21 285 bc3cf17b Sylwester Nawrocki 2019-02-14 286 /* Set capture capability only for boards with the MAX98090 CODEC */ bc3cf17b Sylwester Nawrocki 2019-02-14 287 if (codec_link->num_codecs > 1) { bc3cf17b Sylwester Nawrocki 2019-02-14 288 card->dai_link[0].dpcm_capture = 1; bc3cf17b Sylwester Nawrocki 2019-02-14 289 card->dai_link[1].dpcm_capture = 1; bc3cf17b Sylwester Nawrocki 2019-02-14 290 } a8ad0c85 Sylwester Nawrocki 2017-08-04 291 bc3cf17b Sylwester Nawrocki 2019-02-14 292 priv->sclk_i2s = of_clk_get_by_name(cpu_dai, "i2s_opclk1"); a8ad0c85 Sylwester Nawrocki 2017-08-04 293 if (IS_ERR(priv->sclk_i2s)) { a8ad0c85 Sylwester Nawrocki 2017-08-04 294 ret = PTR_ERR(priv->sclk_i2s); d832d2b2 Sylwester Nawrocki 2019-02-20 295 goto err_put_cpu_dai; a8ad0c85 Sylwester Nawrocki 2017-08-04 296 } a8ad0c85 Sylwester Nawrocki 2017-08-04 297 bc3cf17b Sylwester Nawrocki 2019-02-14 @298 priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis"); a8ad0c85 Sylwester Nawrocki 2017-08-04 299 if (IS_ERR(priv->clk_i2s_bus)) { a8ad0c85 Sylwester Nawrocki 2017-08-04 300 ret = PTR_ERR(priv->clk_i2s_bus); a8ad0c85 Sylwester Nawrocki 2017-08-04 301 goto err_put_sclk; a8ad0c85 Sylwester Nawrocki 2017-08-04 302 } bc3cf17b Sylwester Nawrocki 2019-02-14 303 of_node_put(cpu_dai); a8ad0c85 Sylwester Nawrocki 2017-08-04 304 aba611fc Sylwester Nawrocki 2017-04-21 305 ret = devm_snd_soc_register_card(dev, card); aba611fc Sylwester Nawrocki 2017-04-21 306 if (ret < 0) { aba611fc Sylwester Nawrocki 2017-04-21 307 dev_err(dev, "snd_soc_register_card() failed: %d\n", ret); a8ad0c85 Sylwester Nawrocki 2017-08-04 308 goto err_put_clk_i2s; aba611fc Sylwester Nawrocki 2017-04-21 309 } aba611fc Sylwester Nawrocki 2017-04-21 310 aba611fc Sylwester Nawrocki 2017-04-21 311 return 0; aba611fc Sylwester Nawrocki 2017-04-21 312 a8ad0c85 Sylwester Nawrocki 2017-08-04 313 err_put_clk_i2s: a8ad0c85 Sylwester Nawrocki 2017-08-04 314 clk_put(priv->clk_i2s_bus); a8ad0c85 Sylwester Nawrocki 2017-08-04 315 err_put_sclk: a8ad0c85 Sylwester Nawrocki 2017-08-04 316 clk_put(priv->sclk_i2s); d832d2b2 Sylwester Nawrocki 2019-02-20 317 err_put_cpu_dai: d832d2b2 Sylwester Nawrocki 2019-02-20 318 of_node_put(cpu_dai); bc3cf17b Sylwester Nawrocki 2019-02-14 319 snd_soc_of_put_dai_link_codecs(codec_link); aba611fc Sylwester Nawrocki 2017-04-21 320 return ret; aba611fc Sylwester Nawrocki 2017-04-21 321 } aba611fc Sylwester Nawrocki 2017-04-21 322 :::::: The code at line 298 was first introduced by commit :::::: bc3cf17b575a7a97b4af7ddcf86133175da7a582 ASoC: samsung: odroid: Add support for secondary CPU DAI :::::: TO: Sylwester Nawrocki <s.nawrocki(a)samsung.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: samsung: odroid: Fix of_node refcount unbalance
by Sylwester Nawrocki 20 Feb '19

20 Feb '19

In odroid_audio_probe() some OF nodes are left without reference count decrease after use. Fix it by ensuring required of_node_calls() are done before exiting probe. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index bd2c5163dc7f..c3b0f6c612cb 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -257,27 +257,31 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = of_parse_phandle_with_args(cpu, "sound-dai", "#sound-dai-cells", i, &args); if (ret < 0) - return ret; + break; if (!args.np) { dev_err(dev, "sound-dai property parse error: %d\n", ret); - return -EINVAL; + ret = -EINVAL; + break; } ret = snd_soc_get_dai_name(&args, &link->cpu_dai_name); of_node_put(args.np); if (ret < 0) - return ret; + break; } + if (ret == 0) + cpu_dai = of_parse_phandle(cpu, "sound-dai", 0); - cpu_dai = of_parse_phandle(cpu, "sound-dai", 0); of_node_put(cpu); of_node_put(codec); + if (ret < 0) + return ret; ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link); if (ret < 0) - goto err_put_codec_n; + goto err_put_cpu_dai; /* Set capture capability only for boards with the MAX98090 CODEC */ if (codec_link->num_codecs > 1) { @@ -288,7 +292,7 @@ static int odroid_audio_probe(struct platform_device *pdev) priv->sclk_i2s = of_clk_get_by_name(cpu_dai, "i2s_opclk1"); if (IS_ERR(priv->sclk_i2s)) { ret = PTR_ERR(priv->sclk_i2s); - goto err_put_codec_n; + goto err_put_cpu_dai; } priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis"); @@ -310,7 +314,8 @@ static int odroid_audio_probe(struct platform_device *pdev) clk_put(priv->clk_i2s_bus); err_put_sclk: clk_put(priv->sclk_i2s); -err_put_codec_n: +err_put_cpu_dai: + of_node_put(cpu_dai); snd_soc_of_put_dai_link_codecs(codec_link); return ret; } -- 2.17.1

2 1

[alsa-devel] [PATCH 0/6] ASoC: Fix of-node refcount unbalance
by Takashi Iwai 20 Feb '19

20 Feb '19

Hi, this is a patchset covering the OF-node refcount problem on a few ASoC drivers. thanks, Takashi === Takashi Iwai (6): ASoC: fsl: Fix of-node refcount unbalance in fsl_ssi_probe_from_dt() ASoC: simple-card: Fix missing of_node_put() at simple_dai_link_of() ASoC: simple-card: Fix of-node refcount unbalance in DAI-link parser ASoC: qcom: Fix of-node refcount unbalance in apq8016_sbc_parse_of() ASoC: qcom: Fix of-node refcount unbalance in qcom_snd_parse_of() ASoC: samsung: Fix of-node refcount unbalance in odroid_audio_probe() sound/soc/fsl/fsl_ssi.c | 6 ++++-- sound/soc/generic/simple-card.c | 19 +++++++++++++------ sound/soc/qcom/apq8016_sbc.c | 21 ++++++++++++++++----- sound/soc/qcom/common.c | 10 ++++++++-- sound/soc/samsung/odroid.c | 5 +++-- 5 files changed, 44 insertions(+), 17 deletions(-) -- 2.16.4

5 16

[alsa-devel] [PATCH] ALSA: hda/realtek - Reduce click noise on Dell Precision 5820 headphone
by Takashi Iwai 20 Feb '19

20 Feb '19

Dell Precision 5820 with ALC3234 codec (which is equivalent with ALC255) shows click noises at (runtime) PM resume on the headphone. The biggest source of the noise comes from the cleared headphone pin control at resume, which is done via the standard shutup procedure. Although we have an override of the standard shutup callback to replace with NOP, this would skip other needed stuff (e.g. the pull down of headset power). So, instead, this "fixes" the behavior of alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag. When this flag is set, Realtek codec won't call the standard snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this flag instead of overriding spec->shutup callback itself. This allows us to apply the similar fix for other entries easily if needed in future. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 57 ++++++++++++++++++++++++++----------------- 1 file changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1212955f7ab5..5855648b500c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -118,6 +118,7 @@ struct alc_spec { unsigned int has_alc5505_dsp:1; unsigned int no_depop_delay:1; unsigned int done_hp_init:1; + unsigned int no_shutup_pins:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -476,6 +477,14 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) set_eapd(codec, *p, on); } +static void alc_shutup_pins(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->no_shutup_pins) + snd_hda_shutup_pins(codec); +} + /* generic shutup callback; * just turning off EAPD and a little pause for avoiding pop-noise */ @@ -486,7 +495,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) alc_auto_setup_eapd(codec, false); if (!spec->no_depop_delay) msleep(200); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } /* generic EAPD initialization */ @@ -814,7 +823,7 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); else - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc_reboot_notify(struct hda_codec *codec) @@ -2922,7 +2931,7 @@ static void alc269_shutup(struct hda_codec *codec) (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static struct coef_fw alc282_coefs[] = { @@ -3025,14 +3034,15 @@ static void alc282_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(85); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); alc_write_coef_idx(codec, 0x78, coef78); } @@ -3138,15 +3148,16 @@ static void alc283_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(100); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); alc_write_coef_idx(codec, 0x43, 0x9614); } @@ -3212,14 +3223,15 @@ static void alc256_shutup(struct hda_codec *codec) /* NOTE: call this before clearing the pin, otherwise codec stalls */ alc_update_coef_idx(codec, 0x46, 0, 3 << 12); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc225_init(struct hda_codec *codec) @@ -3306,7 +3318,7 @@ static void alc225_shutup(struct hda_codec *codec) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc_default_init(struct hda_codec *codec) @@ -3360,14 +3372,15 @@ static void alc_default_shutup(struct hda_codec *codec) if (hp_pin_sense) msleep(85); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); if (hp_pin_sense) msleep(100); alc_auto_setup_eapd(codec, false); - snd_hda_shutup_pins(codec); + alc_shutup_pins(codec); } static void alc294_hp_init(struct hda_codec *codec) @@ -3384,8 +3397,9 @@ static void alc294_hp_init(struct hda_codec *codec) msleep(100); - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (!spec->no_shutup_pins) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); alc_update_coef_idx(codec, 0x6f, 0x000f, 0);/* Set HP depop to manual mode */ alc_update_coefex_idx(codec, 0x58, 0x00, 0x8000, 0x8000); /* HP depop procedure start */ @@ -4981,16 +4995,12 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } -static void alc_no_shutup(struct hda_codec *codec) -{ -} - static void alc_fixup_no_shutup(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_PRE_PROBE) { struct alc_spec *spec = codec->spec; - spec->shutup = alc_no_shutup; + spec->no_shutup_pins = 1; } } @@ -6639,6 +6649,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), + SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.16.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Allow compressed buffers in any memory region
by Charles Keepax 20 Feb '19

20 Feb '19

From: Andrew Ford <aford(a)opensource.cirrus.com> Currently, compressed buffers can only be specified in the XM memory region. There is no reason to have such a restriction with the newer meta-data based way of specifying the buffers, so remove it. Signed-off-by: Andrew Ford <aford(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 1dd291cebe67..12ef85e85c29 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -344,6 +344,7 @@ struct wm_adsp_compr_buf { u32 irq_count; int read_index; int avail; + int host_buf_mem_type; }; struct wm_adsp_compr { @@ -3219,14 +3220,14 @@ static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, static inline int wm_adsp_buffer_read(struct wm_adsp_compr_buf *buf, unsigned int field_offset, u32 *data) { - return wm_adsp_read_data_word(buf->dsp, WMFW_ADSP2_XM, + return wm_adsp_read_data_word(buf->dsp, buf->host_buf_mem_type, buf->host_buf_ptr + field_offset, data); } static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, unsigned int field_offset, u32 data) { - return wm_adsp_write_data_word(buf->dsp, WMFW_ADSP2_XM, + return wm_adsp_write_data_word(buf->dsp, buf->host_buf_mem_type, buf->host_buf_ptr + field_offset, data); } @@ -3264,6 +3265,8 @@ static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) if (!buf->host_buf_ptr) return -EIO; + buf->host_buf_mem_type = WMFW_ADSP2_XM; + adsp_dbg(dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); return 0; @@ -3282,6 +3285,7 @@ wm_adsp_find_host_buffer_ctrl(struct wm_adsp_compr_buf *buf) if (!ctl->enabled) continue; + buf->host_buf_mem_type = ctl->alg_region.type; return ctl; } -- 2.11.0

2 4

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Fix a typo in model documentation
by Takashi Iwai 20 Feb '19

20 Feb '19

Some garbage was taken via copy-and-paste error. Clean up. Fixes: a26d96c7802e ("ALSA: hda/realtek - Comprehensive model list for ALC259 & co") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/hd-audio/models.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 368a07a165f5..ef6101a78136 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -254,7 +254,7 @@ alc274-dell-aio ALC274 fixups on Dell AIO machines alc255-dummy-lineout Dell Precision 3930 fixups -alc255-dell-headset"}, +alc255-dell-headset Dell Precision 3630 fixups alc295-hp-x360 HP Spectre X360 fixups -- 2.16.4

1 1

[alsa-devel] [PATCH 0/3] ALSA: Fix of-node refcount unbalance
by Takashi Iwai 19 Feb '19

19 Feb '19

Hi, here is a few patches to address the unbalanced OF-node refcount on powerpc and ac97 driver code. Another patchset for ASoC will follow later. Takashi === Takashi Iwai (3): ALSA: ac97: Fix of-node refcount unbalance ALSA: aoa: Fix of-node refcount unbalance ALSA: ppc: Fix of-node refcount unbalance sound/ac97/bus.c | 2 +- sound/aoa/core/gpio-feature.c | 1 + sound/ppc/tumbler.c | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) -- 2.16.4

3 6

[alsa-devel] [PATCH] ASoC: fsl_spdif: fix sysclk_df type
by Viorel Suman 19 Feb '19

19 Feb '19

According to RM SPDIF STC SYSCLK_DF field is 9-bit wide, values being in 0..511 range. Use a proper type to handle sysclk_df. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a26686e..4842e6d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -96,7 +96,7 @@ struct fsl_spdif_priv { bool dpll_locked; u32 txrate[SPDIF_TXRATE_MAX]; u8 txclk_df[SPDIF_TXRATE_MAX]; - u8 sysclk_df[SPDIF_TXRATE_MAX]; + u16 sysclk_df[SPDIF_TXRATE_MAX]; u8 txclk_src[SPDIF_TXRATE_MAX]; u8 rxclk_src; struct clk *txclk[SPDIF_TXRATE_MAX]; @@ -376,7 +376,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, struct platform_device *pdev = spdif_priv->pdev; unsigned long csfs = 0; u32 stc, mask, rate; - u8 clk, txclk_df, sysclk_df; + u16 sysclk_df; + u8 clk, txclk_df; int ret; switch (sample_rate) { @@ -1109,8 +1110,9 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; - u32 sysclk_dfmin, sysclk_dfmax; - u32 txclk_df, sysclk_df, arate; + u32 arate; + u16 sysclk_dfmin, sysclk_dfmax, sysclk_df; + u8 txclk_df; /* The sysclk has an extra divisor [2, 512] */ sysclk_dfmin = is_sysclk ? 2 : 1; -- 2.7.4

3 3

[alsa-devel] [PATCH] ASoC: fsl_spdif: fix TXCLK_DF mask
by Viorel Suman 19 Feb '19

19 Feb '19

According to RM SPDIF TXCLK_DF mask is 7-bit wide. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_spdif.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 7666dab..e6c61e0 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -152,7 +152,7 @@ enum spdif_gainsel { #define STC_TXCLK_ALL_EN_MASK (1 << STC_TXCLK_ALL_EN_OFFSET) #define STC_TXCLK_ALL_EN (1 << STC_TXCLK_ALL_EN_OFFSET) #define STC_TXCLK_DF_OFFSET 0 -#define STC_TXCLK_DF_MASK (0x7ff << STC_TXCLK_DF_OFFSET) +#define STC_TXCLK_DF_MASK (0x7f << STC_TXCLK_DF_OFFSET) #define STC_TXCLK_DF(x) ((((x) - 1) << STC_TXCLK_DF_OFFSET) & STC_TXCLK_DF_MASK) #define STC_TXCLK_SRC_MAX 8 -- 2.7.4

3 2

[alsa-devel] Applied "ASoC: codecs: pcm186x: Fix energysense SLEEP bit" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: codecs: pcm186x: Fix energysense SLEEP bit 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 05bd7fcdd06b19a10f069af1bea3ad9abac038d7 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Date: Tue, 19 Feb 2019 16:29:28 +0000 Subject: [PATCH] ASoC: codecs: pcm186x: Fix energysense SLEEP bit The ADCs are sleeping when the SLEEP bit is set and running when it's cleared, so the bit should be inverted. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/pcm186x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index c36a391fec8a..c5fcc632f670 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -158,7 +158,7 @@ static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { * Put the codec into SLEEP mode when not in use, allowing the * Energysense mechanism to operate. */ - SND_SOC_DAPM_ADC("ADC", "HiFi Capture", PCM186X_POWER_CTRL, 1, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", PCM186X_POWER_CTRL, 1, 1), }; static const struct snd_soc_dapm_widget pcm1865_dapm_widgets[] = { @@ -184,8 +184,8 @@ static const struct snd_soc_dapm_widget pcm1865_dapm_widgets[] = { * Put the codec into SLEEP mode when not in use, allowing the * Energysense mechanism to operate. */ - SND_SOC_DAPM_ADC("ADC1", "HiFi Capture 1", PCM186X_POWER_CTRL, 1, 0), - SND_SOC_DAPM_ADC("ADC2", "HiFi Capture 2", PCM186X_POWER_CTRL, 1, 0), + SND_SOC_DAPM_ADC("ADC1", "HiFi Capture 1", PCM186X_POWER_CTRL, 1, 1), + SND_SOC_DAPM_ADC("ADC2", "HiFi Capture 2", PCM186X_POWER_CTRL, 1, 1), }; static const struct snd_soc_dapm_route pcm1863_dapm_routes[] = { -- 2.20.1

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: codecs: pcm186x: fix wrong usage of DECLARE_TLV_DB_SCALE()
by Codrin.Ciubotariu＠microchip.com 19 Feb '19

19 Feb '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v2: - fixed title - added correct macro; - revert the removal of a new line; sound/soc/codecs/pcm186x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 809b7e9f03ca..c36a391fec8a 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -42,7 +42,7 @@ struct pcm186x_priv { bool is_master_mode; }; -static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 4000, 50); +static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 50, 0); static const struct snd_kcontrol_new pcm1863_snd_controls[] = { SOC_DOUBLE_R_S_TLV("ADC Capture Volume", PCM186X_PGA_VAL_CH1_L, -- 2.17.1

3 7

[alsa-devel] [PATCH 1/2] ASoC: codecs: pcm186x: fix wrong usage of DECLARE_TLV_DB_LINEAR()
by Codrin.Ciubotariu＠microchip.com 19 Feb '19

19 Feb '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step. Tested on pcm1863. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/pcm186x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index 809b7e9f03ca..7947279243f9 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -6,7 +6,6 @@ * Andreas Dannenberg <dannenberg(a)ti.com> * Andrew F. Davis <afd(a)ti.com> */ - #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -42,7 +41,7 @@ struct pcm186x_priv { bool is_master_mode; }; -static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 4000, 50); +static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 50, 0); static const struct snd_kcontrol_new pcm1863_snd_controls[] = { SOC_DOUBLE_R_S_TLV("ADC Capture Volume", PCM186X_PGA_VAL_CH1_L, -- 2.17.1

1 2

[alsa-devel] [PATCH] ASoC: dapm: Potential small memory leak in dapm_cnew_widget()
by Dan Carpenter 19 Feb '19

19 Feb '19

We should free "w" on the error path. Fixes: 199ed3e81c49 ("ASoC: dapm: fix use-after-free issue with dailink sname") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dea6fc2353e4..1ec06ef6d161 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -332,8 +332,10 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( */ if (_widget->sname) { w->sname = kstrdup_const(_widget->sname, GFP_KERNEL); - if (!w->sname) + if (!w->sname) { + kfree(w); return NULL; + } } return w; } -- 2.17.1

3 2

[alsa-devel] Applied "ASoC: Intel: Headset button support in broxton machine driver" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: Intel: Headset button support in broxton 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 9dd9b210f8c6104690ba48a630bbe9af2f32c292 Mon Sep 17 00:00:00 2001 From: Yong Zhi <yong.zhi(a)intel.com> Date: Wed, 13 Feb 2019 17:08:51 -0600 Subject: [PATCH] ASoC: Intel: Headset button support in broxton machine driver Map the 4 headset buttons to KEY_PAUSE, KEY_VOLUMEUP, KEY_VOLUMEDOWN and KEY_VOICECOMMAND. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index c00925f9da73..30311b81a543 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -16,6 +16,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -193,6 +194,12 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_3, + KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &broxton_headset); snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); -- 2.20.1

1 0

[alsa-devel] [PATCH v4 1/3] ASoC: Intel: Headset button support in broxton machine driver
by Yong Zhi 19 Feb '19

19 Feb '19

Map the 4 headset buttons to KEY_PAUSE, KEY_VOLUMEUP, KEY_VOLUMEDOWN and KEY_VOICECOMMAND. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- Changes for v4: - Rebased on the latest for-next branch. sound/soc/intel/boards/bxt_da7219_max98357a.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index c00925f9da73..30311b81a543 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -16,6 +16,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -193,6 +194,12 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_3, + KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &broxton_headset); snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); -- 2.7.4

2 3

[alsa-devel] Applied "ASoC: Intel: glk: Add DAI links for Multi-Playback" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: Intel: glk: Add DAI links for Multi-Playback 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 bc3523a3acb3ba311d5d9939901ff2b7f8833e44 Mon Sep 17 00:00:00 2001 From: Yong Zhi <yong.zhi(a)intel.com> Date: Wed, 13 Feb 2019 17:08:53 -0600 Subject: [PATCH] ASoC: Intel: glk: Add DAI links for Multi-Playback Add FE DAI link to support parallel playback on 2 ports simultaneously. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 407b0cfc5167..5cadb7f654f3 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -51,6 +51,7 @@ struct bxt_card_private { enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, + BXT_DPCM_AUDIO_HS_PB, BXT_DPCM_AUDIO_REF_CP, BXT_DPCM_AUDIO_DMIC_CP, BXT_DPCM_AUDIO_HDMI1_PB, @@ -405,6 +406,20 @@ static struct snd_soc_dai_link broxton_dais[] = { .dpcm_capture = 1, .ops = &broxton_da7219_fe_ops, }, + [BXT_DPCM_AUDIO_HS_PB] = { + .name = "Bxt Audio Headset Playback", + .stream_name = "Headset Playback", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .ops = &broxton_da7219_fe_ops, + }, [BXT_DPCM_AUDIO_REF_CP] = { .name = "Bxt Audio Reference cap", -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Fix secondary platform device unregistration" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: samsung: i2s: Fix secondary platform device unregistration 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 022c4156697b9ae30a00f5cd7cee08ed61554e86 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Tue, 19 Feb 2019 16:19:40 +0100 Subject: [PATCH] ASoC: samsung: i2s: Fix secondary platform device unregistration This fixes unregistration of the secondary platform device so all resources are properly released. Additionally the removal sequence is corrected so it is in reverse order comparing to probe sequence. The test against NULL priv->pdev_sec is removed as it is not necessary. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 02472f576e17..cd92bb6e1da1 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1359,11 +1359,10 @@ static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) static void i2s_delete_secondary_device(struct samsung_i2s_priv *priv) { - if (priv->pdev_sec) { - platform_device_del(priv->pdev_sec); - priv->pdev_sec = NULL; - } + platform_device_unregister(priv->pdev_sec); + priv->pdev_sec = NULL; } + static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; @@ -1487,14 +1486,14 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai->filter, "tx-sec", NULL, &pdev->dev); if (ret < 0) - goto err_disable_clk; + goto err_del_sec; } if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); ret = -EINVAL; - goto err_disable_clk; + goto err_del_sec; } dev_set_drvdata(&pdev->dev, priv); @@ -1503,7 +1502,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) &samsung_i2s_component, priv->dai_drv, num_dais); if (ret < 0) - goto err_disable_clk; + goto err_del_sec; pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); @@ -1518,9 +1517,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) err_disable_pm: pm_runtime_disable(&pdev->dev); +err_del_sec: + i2s_delete_secondary_device(priv); err_disable_clk: clk_disable_unprepare(priv->clk); - i2s_delete_secondary_device(priv); return ret; } @@ -1536,9 +1536,10 @@ static int samsung_i2s_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); i2s_unregister_clock_provider(priv); + i2s_delete_secondary_device(priv); clk_disable_unprepare(priv->clk); + pm_runtime_put_noidle(&pdev->dev); - i2s_delete_secondary_device(priv); return 0; } -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization 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 c6bebefa2f0603fb21ae329521e15461b0486679 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Tue, 19 Feb 2019 16:19:41 +0100 Subject: [PATCH] ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio interfaces" and the driver will try to register there multiple times same platform device for the secondary FIFO, which of course fails miserably. To fix this we derive the secondary platform device name from the primary device name. The secondary device name will now be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec". The fixed platform_device_id table entry is removed as the secondary device name is now dynamic and device/driver matching is done through driver_override. Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Suggested-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 50 +++++++++++++++++++++++++------------- sound/soc/samsung/odroid.c | 2 +- 2 files changed, 34 insertions(+), 18 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index cd92bb6e1da1..4231001226f4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1339,20 +1339,35 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) /* Create platform device for the secondary PCM */ static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) { - struct platform_device *pdev; + struct platform_device *pdev_sec; + const char *devname; int ret; - pdev = platform_device_register_simple("samsung-i2s-sec", -1, NULL, 0); - if (!pdev) + devname = devm_kasprintf(&priv->pdev->dev, GFP_KERNEL, "%s-sec", + dev_name(&priv->pdev->dev)); + if (!devname) return -ENOMEM; - ret = device_attach(&pdev->dev); + pdev_sec = platform_device_alloc(devname, -1); + if (!pdev_sec) + return -ENOMEM; + + pdev_sec->driver_override = kstrdup("samsung-i2s", GFP_KERNEL); + + ret = platform_device_add(pdev_sec); if (ret < 0) { - dev_info(&pdev->dev, "device_attach() failed\n"); + platform_device_put(pdev_sec); return ret; } - priv->pdev_sec = pdev; + ret = device_attach(&pdev_sec->dev); + if (ret <= 0) { + platform_device_unregister(priv->pdev_sec); + dev_info(&pdev_sec->dev, "device_attach() failed\n"); + return ret; + } + + priv->pdev_sec = pdev_sec; return 0; } @@ -1367,22 +1382,25 @@ static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; - struct resource *res; u32 regs_base, idma_addr = 0; struct device_node *np = pdev->dev.of_node; const struct samsung_i2s_dai_data *i2s_dai_data; - int num_dais, ret; + const struct platform_device_id *id; struct samsung_i2s_priv *priv; + struct resource *res; + int num_dais, ret; - if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) + if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { i2s_dai_data = of_device_get_match_data(&pdev->dev); - else - i2s_dai_data = (struct samsung_i2s_dai_data *) - platform_get_device_id(pdev)->driver_data; + } else { + id = platform_get_device_id(pdev); - /* Nothing to do if it is the secondary device probe */ - if (!i2s_dai_data) - return 0; + /* Nothing to do if it is the secondary device probe */ + if (!id) + return 0; + + i2s_dai_data = (struct samsung_i2s_dai_data *)id->driver_data; + } priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -1637,8 +1655,6 @@ static const struct platform_device_id samsung_i2s_driver_ids[] = { { .name = "samsung-i2s", .driver_data = (kernel_ulong_t)&i2sv3_dai_type, - }, { - .name = "samsung-i2s-sec", }, {}, }; diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 5b2bcd1d3450..bd2c5163dc7f 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -185,7 +185,7 @@ static struct snd_soc_dai_link odroid_card_dais[] = { .ops = &odroid_card_fe_ops, .name = "Secondary", .stream_name = "Secondary", - .platform_name = "samsung-i2s-sec", + .platform_name = "3830000.i2s-sec", .dynamic = 1, .dpcm_playback = 1, } -- 2.20.1

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: samsung: i2s: Fix secondary platform device unregistration
by Sylwester Nawrocki 19 Feb '19

19 Feb '19

This fixes unregistration of the secondary platform device so all resources are properly released. Additionally the removal sequence is corrected so it is in reverse order comparing to probe sequence. The test against NULL priv->pdev_sec is removed as it is not necessary. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v1: - corrected order of calls on error path in probe() and in remove(). sound/soc/samsung/i2s.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 02472f576e17..cd92bb6e1da1 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1359,11 +1359,10 @@ static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) static void i2s_delete_secondary_device(struct samsung_i2s_priv *priv) { - if (priv->pdev_sec) { - platform_device_del(priv->pdev_sec); - priv->pdev_sec = NULL; - } + platform_device_unregister(priv->pdev_sec); + priv->pdev_sec = NULL; } + static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; @@ -1487,14 +1486,14 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai->filter, "tx-sec", NULL, &pdev->dev); if (ret < 0) - goto err_disable_clk; + goto err_del_sec; } if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); ret = -EINVAL; - goto err_disable_clk; + goto err_del_sec; } dev_set_drvdata(&pdev->dev, priv); @@ -1503,7 +1502,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) &samsung_i2s_component, priv->dai_drv, num_dais); if (ret < 0) - goto err_disable_clk; + goto err_del_sec; pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); @@ -1518,9 +1517,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) err_disable_pm: pm_runtime_disable(&pdev->dev); +err_del_sec: + i2s_delete_secondary_device(priv); err_disable_clk: clk_disable_unprepare(priv->clk); - i2s_delete_secondary_device(priv); return ret; } @@ -1536,9 +1536,10 @@ static int samsung_i2s_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); i2s_unregister_clock_provider(priv); + i2s_delete_secondary_device(priv); clk_disable_unprepare(priv->clk); + pm_runtime_put_noidle(&pdev->dev); - i2s_delete_secondary_device(priv); return 0; } -- 2.17.1

2 3

[alsa-devel] [PATCH] ASoC: topology: free created components in tplg load error
by Bard liao 19 Feb '19

19 Feb '19

Topology resources are no longer needed if any element failed to load. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-topology.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c5638e15a2dd..25fca7055464 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2587,6 +2587,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id) { struct soc_tplg tplg; + int ret; /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); @@ -2600,7 +2601,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.bytes_ext_ops = ops->bytes_ext_ops; tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; - return soc_tplg_load(&tplg); + ret = soc_tplg_load(&tplg); + /* free the created components if fail to load topology */ + if (ret) + snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); -- 2.17.1

3 3

[alsa-devel] [GIT PULL] ASoC fixes for v5.0
by Mark Brown 19 Feb '19

19 Feb '19

The following changes since commit 860b454c2c0cbda6892954f5cdbbb48931b3c8db: ASoC: samsung: Prevent clk_get_rate() calls in atomic context (2019-02-07 14:34:53 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.0-rc6 for you to fetch changes up to 19dd0777773ab17b4d97f7105e836867c0cdecb4: ASoC: simple-card: fixup refcount_t underflow (2019-02-18 18:36:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.0 A few small fixes, a driver fix for Samsung, a fix for refcounting of of_nodes in the simple-card driver that triggered on a lot of systems and a fix for topology error handling. ---------------------------------------------------------------- Bard liao (1): ASoC: topology: free created components in tplg load error Kuninori Morimoto (1): ASoC: simple-card: fixup refcount_t underflow Sylwester Nawrocki (1): ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI sound/soc/generic/simple-card.c | 2 +- sound/soc/samsung/i2s.c | 8 +++++++- sound/soc/soc-topology.c | 8 +++++++- 3 files changed, 15 insertions(+), 3 deletions(-)

2 1

[alsa-devel] [bug report] ASoC: dapm: Consolidate input and output path handling
by Dan Carpenter 19 Feb '19

19 Feb '19

Hello Lars-Peter Clausen, The patch a3423b02cf74: "ASoC: dapm: Consolidate input and output path handling" from Aug 11, 2015, leads to the following static checker warning: sound/soc/soc-dapm.c:2831 snd_soc_dapm_add_path() error: potential zalloc NULL dereference: 'path->sink' sound/soc/soc-dapm.c 2791 2792 path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL); 2793 if (!path) 2794 return -ENOMEM; 2795 2796 path->node[SND_SOC_DAPM_DIR_IN] = wsource; 2797 path->node[SND_SOC_DAPM_DIR_OUT] = wsink; We used to say "path->sink = wsink;" here, but now it's NULL from the kzalloc(). 2798 widgets[SND_SOC_DAPM_DIR_IN] = wsource; 2799 widgets[SND_SOC_DAPM_DIR_OUT] = wsink; 2800 2801 path->connected = connected; 2802 INIT_LIST_HEAD(&path->list); 2803 INIT_LIST_HEAD(&path->list_kcontrol); 2804 2805 if (wsource->is_supply || wsink->is_supply) 2806 path->is_supply = 1; 2807 2808 /* connect static paths */ 2809 if (control == NULL) { 2810 path->connect = 1; 2811 } else { 2812 switch (wsource->id) { 2813 case snd_soc_dapm_demux: 2814 ret = dapm_connect_mux(dapm, path, control, wsource); 2815 if (ret) 2816 goto err; 2817 break; 2818 default: 2819 break; 2820 } 2821 2822 switch (wsink->id) { 2823 case snd_soc_dapm_mux: 2824 ret = dapm_connect_mux(dapm, path, control, wsink); 2825 if (ret != 0) 2826 goto err; 2827 break; 2828 case snd_soc_dapm_switch: 2829 case snd_soc_dapm_mixer: 2830 case snd_soc_dapm_mixer_named_ctl: --> 2831 ret = dapm_connect_mixer(dapm, path, control); ^^^^ Dereferenced inside the function. It's hard to believe that this has been broken for three years, but I can't see another explanation. (0)_(0)!!! 2832 if (ret != 0) 2833 goto err; 2834 break; 2835 default: 2836 break; 2837 } 2838 } 2839 2840 list_add(&path->list, &dapm->card->paths); 2841 snd_soc_dapm_for_each_direction(dir) regards, dan carpenter

1 0

[alsa-devel] [PATCH 1/3] SoC: imx-sgtl5000: add missing put_device()
by Wen Yang 19 Feb '19

19 Feb '19

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 --- 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 b6cb80480b60..bf8597f57dce 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_dbg(&pdev->dev, "failed to find codec platform device\n"); -- 2.20.1

2 1

[alsa-devel] Applied "ASoC: adau1977: Fix reset-gpios typo" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: adau1977: Fix reset-gpios typo 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 d98afe1353b6dbd1527f3833da70d8525e6dbae3 Mon Sep 17 00:00:00 2001 From: Bogdan Togorean <bogdan.togorean(a)analog.com> Date: Fri, 15 Feb 2019 12:26:33 +0200 Subject: [PATCH] ASoC: adau1977: Fix reset-gpios typo This change fixes a typo in the dt-binding examples (reset_gpio -> reset-gpios). Even though 'reset-gpio' is a valid construct for gpiolib the naming 'reset-gpios' is more suited for dt-bindings documentation. Signed-off-by: Bogdan Togorean <bogdan.togorean(a)analog.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/adi,adau1977.txt | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt index e79aeef73f28..11acd94b1665 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.txt @@ -17,7 +17,7 @@ Required properties: Documentation/devicetree/bindings/regulator/regulator.txt Optional properties: - - reset-gpio: the reset pin for the chip, for more details consult + - reset-gpios: the reset pin for the chip, for more details consult Documentation/devicetree/bindings/gpio/gpio.txt - DVDD-supply: supply voltage for the digital core, please consult @@ -40,7 +40,7 @@ Examples: AVDD-supply = <&regulator>; DVDD-supply = <&regulator_digital>; - reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; }; adau1977_i2c: adau1977@11 { @@ -50,5 +50,5 @@ Examples: AVDD-supply = <&regulator>; DVDD-supply = <&regulator_digital>; - reset_gpio = <&gpio 10 GPIO_ACTIVE_LOW>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; }; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: fsl_esai: fix register setting issue in RIGHT_J mode" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: fsl_esai: fix register setting issue in RIGHT_J mode 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 cc29ea007347f39f4c5a4d27b0b555955a0277f9 Mon Sep 17 00:00:00 2001 From: "S.j. Wang" <shengjiu.wang(a)nxp.com> Date: Mon, 18 Feb 2019 08:29:11 +0000 Subject: [PATCH] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 57b484768a58..afe67c865330 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: wm8741: Make function 'wm8741_mute' static" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: wm8741: Make function 'wm8741_mute' static 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 76a60f312f64dc48450b15a7f167b46e6230e4d1 Mon Sep 17 00:00:00 2001 From: Wei Yongjun <weiyongjun1(a)huawei.com> Date: Sat, 16 Feb 2019 01:35:56 +0000 Subject: [PATCH] ASoC: wm8741: Make function 'wm8741_mute' static Fixes the following sparse warning: sound/soc/codecs/wm8741.c:371:5: warning: symbol 'wm8741_mute' was not declared. Should it be static? Fixes: 36b1599340b5 ("ASoC: wm8741: Add digital mute callback") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm8741.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a4b8c459ea57..546ea735f534 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -368,7 +368,7 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) +static int wm8741_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_component *component = codec_dai->component; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: samsung: odroid: Add missing DAPM routes" to the asoc tree
by Mark Brown 19 Feb '19

19 Feb '19

The patch ASoC: samsung: odroid: Add missing DAPM routes 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 f89aea0f132142b29dc0c8cf4d445bd12db7b1a6 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Fri, 15 Feb 2019 13:04:22 +0100 Subject: [PATCH] ASoC: samsung: odroid: Add missing DAPM routes With old DTS there will be missing DAPM routes linking BE with CODECs. Add those routes in the card driver so sound works properly on Odroid XU3/4 also without DTS updates enabling the secondary PCM. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/odroid.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 941e8c3f67a4..5b2bcd1d3450 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -149,6 +149,12 @@ static const struct snd_soc_ops odroid_card_be_ops = { .trigger = odroid_card_be_trigger, }; +/* DAPM routes for backward compatibility with old DTS */ +static const struct snd_soc_dapm_route odroid_dapm_routes[] = { + { "I2S Playback", NULL, "Mixer DAI TX" }, + { "HiFi Playback", NULL, "Mixer DAI TX" }, +}; + static struct snd_soc_dai_link odroid_card_dais[] = { { /* Primary FE <-> BE link */ @@ -237,11 +243,15 @@ static int odroid_audio_probe(struct platform_device *pdev) /* * For backwards compatibility create the secondary CPU DAI link only * if there are 2 CPU DAI entries in the cpu sound-dai property in DT. + * Also add required DAPM routes not available in old DTS. */ num_pcms = of_count_phandle_with_args(cpu, "sound-dai", "#sound-dai-cells"); - if (num_pcms == 1) + if (num_pcms == 1) { + card->dapm_routes = odroid_dapm_routes; + card->num_dapm_routes = ARRAY_SIZE(odroid_dapm_routes); card->num_links--; + } for (i = 0; i < num_pcms; i++, link += 2) { ret = of_parse_phandle_with_args(cpu, "sound-dai", -- 2.20.1

1 0

Re: [alsa-devel] [PATCH] dt-bindings: adau1977: Fix reset-gpios typo
by Mark Brown 19 Feb '19

19 Feb '19

On Fri, Feb 15, 2019 at 12:26:33PM +0200, Bogdan Togorean wrote: > This change fixes a typo in the dt-binding examples (reset_gpio -> > reset-gpios). > Even though 'reset-gpio' is a valid construct for gpiolib the naming > 'reset-gpios' is more suited for dt-bindings documentation. Please use subject lines matching the style for the subsystem. This makes it easier for people to identify relevant patches.

1 0

[alsa-devel] Can't load alsa-project.org
by 積丹尼 Dan Jacobson 19 Feb '19

19 Feb '19

Dear ALSA team, you need to update the TLS on alsa-project.org > That's because the site apparently only offers TLS 1.0 which libssl > does not support anymore by default. See https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=921523 P.S., The side panel of https://alsa-project.org/ has a link to "Bug tracking" which is broken and needs to link to https://alsa-project.org/main/index.php/Bug_Tracking instead.

2 1

[alsa-devel] [PATCH] ALSA: xen-front: remove unneeded switch fall-through
by Li RongQing 19 Feb '19

19 Feb '19

This case block will be terminated by a break, so not need a switch fall-through Signed-off-by: Li RongQing <lirongqing(a)baidu.com> --- sound/xen/xen_snd_front.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c index a9e5c2cd7698..f12fb8eeaa27 100644 --- a/sound/xen/xen_snd_front.c +++ b/sound/xen/xen_snd_front.c @@ -250,7 +250,6 @@ static void sndback_changed(struct xenbus_device *xb_dev, case XenbusStateReconfigured: /* fall through */ case XenbusStateInitialised: - /* fall through */ break; case XenbusStateInitialising: -- 2.16.2

2 1

[alsa-devel] [PATCH v2 00/21] ASoC: dmaengine updates, secondary CPU DAI for Odroid boards
by Sylwester Nawrocki 18 Feb '19

18 Feb '19

This series restores support for the secondary CPU DAI in samsung/i2s.c driver and adds support for secondary CPU DAI on Odroid boards. I have added 4 more clean up patches comparing to first version of the patch set (16/22...19/22). The first patch adds new platform device for the secondary PCM, I tried to avoid registering new platform device in probe() as it doesn't play well with deferred probing but couldn't find a way to avoid that new device, unique devices are needed to register "PCM" components for each CPU DAI. Patches 2/21...11/21 are mostly a clean up, following conversion to single component with multiple DAIs, allowing to reference both primary/secondary DAI in DT. The patch set has been tested on Odroid U3, XU3, XU4 and Mini2440 (s3c2440 non-DT) boards. Changes since v1 ((addressing Krzysztof's review comments): - added a comment, whitespace change in patch 1/21 - dropped unnecessary new line addition from patch 1/21 Sylwester Nawrocki (21): ASoC: samsung: i2s: Restore support for the secondary PCM ASoC: samsung: i2s: Move clk supplier data to common driver data structure ASoC: samsung: i2s: Add widgets and routes for DPCM support ASoC: samsung: i2s: Move core clk to the driver common data structure ASoC: samsung: i2s: Move opclk data to common driver data structure ASoC: samsung: i2s: Move registers cache to common driver data structure ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: Drop DAI DMA data chan_name assignments ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition ASoC: samsung: i2s: Comments clean up ASoC: samsung: i2s: Convert to SPDX License Indentifier ARM: dts: exynos5422-odroidxu3: Add support for secondary DAI ARM: dts: exynos5422-odroidxu4: Add support for secondary DAI .../boot/dts/exynos5422-odroidxu3-audio.dtsi | 6 +- arch/arm/boot/dts/exynos5422-odroidxu4.dts | 4 +- include/sound/dmaengine_pcm.h | 6 - sound/soc/samsung/dmaengine.c | 12 +- sound/soc/samsung/i2s.c | 543 ++++++++++-------- sound/soc/samsung/odroid.c | 131 +++-- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 21 +- 9 files changed, 413 insertions(+), 318 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: qcom: Kconfig: select dmic for sdm845
by Cheng-Yi Chiang 18 Feb '19

18 Feb '19

sdm845 uses dmic on EC so it should select CROS_EC_CODEC. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 7948e993adba..8f206cb4fcc0 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -103,6 +103,7 @@ config SND_SOC_SDM845 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 + select SND_SOC_CROS_EC_CODEC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.21.0.rc0.258.g878e2cd30e-goog

2 2

[alsa-devel] [PATCH -next] ASoC: cs35l36: Make some symbols static
by Wei Yongjun 18 Feb '19

18 Feb '19

Fixes the following sparse warnings: sound/soc/codecs/cs35l36.c:135:20: warning: symbol 'cs35l36_reg' was not declared. Should it be static? sound/soc/codecs/cs35l36.c:248:6: warning: symbol 'cs35l36_readable_reg' was not declared. Should it be static? sound/soc/codecs/cs35l36.c:398:6: warning: symbol 'cs35l36_precious_reg' was not declared. Should it be static? sound/soc/codecs/cs35l36.c:410:6: warning: symbol 'cs35l36_volatile_reg' was not declared. Should it be static? Fixes: 6ba9dd6c893b ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/cs35l36.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index 4f880a678812..005f1adc80b8 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -132,7 +132,7 @@ static const struct cs35l36_pll_config cs35l36_pll_sysclk[] = { {27000000, 0x3F, 0x0A}, }; -struct reg_default cs35l36_reg[] = { +static struct reg_default cs35l36_reg[] = { {CS35L36_TESTKEY_CTRL, 0x00000000}, {CS35L36_USERKEY_CTL, 0x00000000}, {CS35L36_OTP_CTRL1, 0x00002460}, @@ -245,7 +245,7 @@ struct reg_default cs35l36_reg[] = { {CS35L36_PAC_INT7_CTRL, 0x00000001}, }; -bool cs35l36_readable_reg(struct device *dev, unsigned int reg) +static bool cs35l36_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L36_SW_RESET: @@ -395,7 +395,7 @@ bool cs35l36_readable_reg(struct device *dev, unsigned int reg) } } -bool cs35l36_precious_reg(struct device *dev, unsigned int reg) +static bool cs35l36_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L36_TESTKEY_CTRL: @@ -407,7 +407,7 @@ bool cs35l36_precious_reg(struct device *dev, unsigned int reg) } } -bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) +static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case CS35L36_SW_RESET:

3 3

[alsa-devel] [PATCH -next] ASoC: stm32: sai: remove set but not used variables 'mask, cr1'
by YueHaibing 18 Feb '19

18 Feb '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_configure_clock': sound/soc/stm/stm32_sai_sub.c:902:11: warning: variable 'mask' set but not used [-Wunused-but-set-variable] sound/soc/stm/stm32_sai_sub.c:902:6: warning: variable 'cr1' set but not used [-Wunused-but-set-variable] It's not used any more after 8307b2afd386 ("ASoC: stm32: sai: set sai as mclk clock provider") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/stm/stm32_sai_sub.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d4825700b63f..a00640f1df06 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -898,7 +898,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, struct snd_pcm_hw_params *params) { struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai); - int cr1, mask, div = 0; + int div = 0; int sai_clk_rate, mclk_ratio, den; unsigned int rate = params_rate(params); @@ -944,8 +944,7 @@ static int stm32_sai_configure_clock(struct snd_soc_dai *cpu_dai, if (sai->mclk_rate) { mclk_ratio = sai->mclk_rate / rate; if (mclk_ratio == 512) { - mask = SAI_XCR1_OSR; - cr1 = SAI_XCR1_OSR; + ; } else if (mclk_ratio != 256) { dev_err(cpu_dai->dev, "Wrong mclk ratio %d\n",

3 4

[alsa-devel] [PATCH] ASoC: samsung: odroid: Ensure proper sample rate on pri/sec PCM
by Sylwester Nawrocki 18 Feb '19

18 Feb '19

Currently when playing sound with different sample rates actual sample rate will be determined by audio stream which starts first on either primary or secondary PCM. The audio root clock will be configured appropriately only for the first stream. As the hardware is limited to same sample rate on both interfaces we need to disallow streams with different sample rates. It is done by this patch by returning error in FE hw_params if there is already active stream running with different sample rate. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 56 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 18bb3bfe0300..941e8c3f67a4 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -20,6 +20,11 @@ struct odroid_priv { struct snd_soc_card card; struct clk *clk_i2s_bus; struct clk *sclk_i2s; + + /* Spinlock protecting fields below */ + spinlock_t lock; + unsigned int be_sample_rate; + bool be_active; }; static int odroid_card_fe_startup(struct snd_pcm_substream *substream) @@ -31,8 +36,25 @@ static int odroid_card_fe_startup(struct snd_pcm_substream *substream) return 0; } +static int odroid_card_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); + unsigned long flags; + int ret = 0; + + spin_lock_irqsave(&priv->lock, flags); + if (priv->be_active && priv->be_sample_rate != params_rate(params)) + ret = -EINVAL; + spin_unlock_irqrestore(&priv->lock, flags); + + return ret; +} + static const struct snd_soc_ops odroid_card_fe_ops = { .startup = odroid_card_fe_startup, + .hw_params = odroid_card_fe_hw_params, }; static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, @@ -41,6 +63,7 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); unsigned int pll_freq, rclk_freq, rfs; + unsigned long flags; int ret; switch (params_rate(params)) { @@ -87,11 +110,43 @@ static int odroid_card_be_hw_params(struct snd_pcm_substream *substream, return ret; } + spin_lock_irqsave(&priv->lock, flags); + priv->be_sample_rate = params_rate(params); + spin_unlock_irqrestore(&priv->lock, flags); + + return 0; +} + +static int odroid_card_be_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); + unsigned long flags; + + spin_lock_irqsave(&priv->lock, flags); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->be_active = true; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->be_active = false; + break; + } + + spin_unlock_irqrestore(&priv->lock, flags); + return 0; } static const struct snd_soc_ops odroid_card_be_ops = { .hw_params = odroid_card_be_hw_params, + .trigger = odroid_card_be_trigger, }; static struct snd_soc_dai_link odroid_card_dais[] = { @@ -150,6 +205,7 @@ static int odroid_audio_probe(struct platform_device *pdev) card->owner = THIS_MODULE; card->fully_routed = true; + spin_lock_init(&priv->lock); snd_soc_card_set_drvdata(card, priv); ret = snd_soc_of_parse_card_name(card, "model"); -- 2.17.1

3 3

[alsa-devel] [PATCH] ASoC: simple-card: Fix refcount underflow
by Daniel Baluta 18 Feb '19

18 Feb '19

of_get_child_by_name() takes a reference we'll need to drop later so when we substitute in top we need to take a reference as well as just assigning. Without this patch we hit the following error: [ 1.246852] OF: ERROR: Bad of_node_put() on /sound-wm8524 [ 1.262261] Hardware name: NXP i.MX8MQ EVK (DT) [ 1.266807] Workqueue: events deferred_probe_work_func [ 1.271950] Call trace: [ 1.274406] dump_backtrace+0x0/0x158 [ 1.278074] show_stack+0x14/0x20 [ 1.281396] dump_stack+0xa8/0xcc [ 1.284717] of_node_release+0xb0/0xc8 [ 1.288474] kobject_put+0x74/0xf0 [ 1.291879] of_node_put+0x14/0x28 [ 1.295286] __of_get_next_child+0x44/0x70 [ 1.299387] of_get_next_child+0x3c/0x60 [ 1.303315] simple_for_each_link+0x1dc/0x230 [ 1.307676] simple_probe+0x80/0x540 [ 1.311256] platform_drv_probe+0x50/0xa0 This patch is based on an earlier version posted by Kuninori Morimoto and commit message includes explanations from Mark Brown. https://patchwork.kernel.org/patch/10814255/ Reported-by: Vicente Bergas <vicencb(a)gmail.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 08df261024cf..dc18c4492955 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -445,7 +445,7 @@ static int simple_for_each_link(struct simple_priv *priv, /* Check if it has dai-link */ node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { - node = top; + node = of_node_get(top); is_top = 1; } -- 2.17.1

5 9

[alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Kuninori Morimoto 18 Feb '19

18 Feb '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") merged simple-card and simple-scu-card. Then it had refcount underflow bug. This patch fixup it. We will get below error without this patch. OF: ERROR: Bad of_node_put() on /sound CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514 Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) Workqueue: events deferred_probe_work_func Call trace: dump_backtrace+0x0/0x150 show_stack+0x24/0x30 dump_stack+0xb0/0xec of_node_release+0xd0/0xd8 kobject_put+0x74/0xe8 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x40/0x68 asoc_simple_card_probe+0x604/0x730 platform_drv_probe+0x58/0xa8 ... Reported-by: Vicente Bergas <vicencb(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Vicente, can you please test this patch ? sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 37e001c..3fe3441 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -462,7 +462,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) conf_idx = 0; node = of_get_child_by_name(top, PREFIX "dai-link"); if (!node) { - node = dev->of_node; + node = of_node_get(top); loop = 0; } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsi: fix spelling mistake "doens't" -> "doesn't"
by Colin King 18 Feb '19

18 Feb '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 285afbafa662..3447dbdba1f1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -780,7 +780,7 @@ static int fsi_clk_init(struct device *dev, return -EINVAL; } if (clock->div == clock->own) { - dev_err(dev, "cpu doens't support div clock\n"); + dev_err(dev, "cpu doesn't support div clock\n"); return -EINVAL; } } -- 2.20.1

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[alsa-devel] [PATCH 0/5] ASoC: codecs: ad193x: Several fixes and imprevements
by Codrin.Ciubotariu＠microchip.com 18 Feb '19

18 Feb '19

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> This patch set contains some changes I needed to make I2S and TDM (DSP_A) formats work. I tested only with ad1936, but the patches should work fine for all the codecs that use this driver. Although I checked the DS for all of them (I hope) to assure that the patches apply, I would really appreciate if someone else can test them on another codec. Codrin Ciubotariu (5): ASoC: codecs: ad193x: Remove capture support for codecs without ADC ASoC: codecs: ad193x: Set constraint to always have 32 sample bits ASoC: codecs: ad193x: Fix frame polarity for DSP_A format ASoC: codecs: ad193x: Add runtime support for DSP_A and I2S modes ASoC: codecs: ad193x: Add support to disable on-chip PLL sound/soc/codecs/ad193x.c | 76 ++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/ad193x.h | 8 +++++ 2 files changed, 80 insertions(+), 4 deletions(-) -- 2.17.1

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[alsa-devel] snd_hda_codec_hdmi: `hdaudio hdaudioC0D2: Unable to bind the codec`
by Paul Menzel 18 Feb '19

18 Feb '19

Dear Linux folks, With drm-tip (v4.19-rc3-946-g09b295662edd) Linux prints `snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec` to the log. ``` […] [ 12.481788] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC671: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 12.482536] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=1 (0x17/0x0/0x0/0x0/0x0) [ 12.483184] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 12.483737] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 12.484198] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 12.484582] snd_hda_codec_realtek hdaudioC0D0: Front Mic=0x19 [ 12.485100] snd_hda_codec_realtek hdaudioC0D0: Rear Mic=0x18 [ 12.485519] snd_hda_codec_realtek hdaudioC0D0: Line=0x1a [ 12.497685] snd_hda_codec_hdmi hdaudioC0D2: No i915 binding for Intel HDMI/DP codec [ 12.498311] hdaudio hdaudioC0D2: Unable to bind the codec [ 12.498789] input: HDA Intel PCH Front Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input6 [ 12.499567] input: HDA Intel PCH Rear Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input7 [ 12.500268] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1f.3/sound/card0/input8 [ 12.501039] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1f.3/sound/card0/input9 [ 12.501683] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input10 [ 17.743338] [drm:intel_pch_type [i915]] Found SunrisePoint PCH […] ``` Is that an error or just a notice? Kind regards, Paul

2 12

Re: [alsa-devel] [PATCH 1/2] ASoC: samsung: i2s: Fix the secondary platform device registration
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Fri, 15 Feb 2019 at 15:48, Sylwester Nawrocki <s.nawrocki(a)samsung.com> wrote: > > This fixes unregistration of the secondary platform device so all > resources are properly released. The test for NULL priv->pdev_sec > is not necessary and it is removed. > > Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > --- > This patch is based off of ASoC for-next and patches: > ASoC: samsung: odroid: Ensure proper sample rate on pri/sec PCM > ASoC: samsung: i2s: Prevent potential NULL platform data dereference > ASoC: samsung: odroid: Add missing DAPM routes > > sound/soc/samsung/i2s.c | 7 +++---- > 1 file changed, 3 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c > index 6bf0f55d1e51..e36c44e2f1bb 100644 > --- a/sound/soc/samsung/i2s.c > +++ b/sound/soc/samsung/i2s.c > @@ -1359,11 +1359,10 @@ static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) > > static void i2s_delete_secondary_device(struct samsung_i2s_priv *priv) > { > - if (priv->pdev_sec) { > - platform_device_del(priv->pdev_sec); > - priv->pdev_sec = NULL; > - } > + platform_device_unregister(priv->pdev_sec); > + priv->pdev_sec = NULL; Reviewed-by: Krzysztof Kozlowski <krzk(a)kernel.org> Although I think that you might need to re-order calls in samsung_i2s_remove(). In general they should be in exact reverse order of probe(). In this case, clk_disable_unprepare(priv->clk) should be after unregistering secondary device. If order has to be different because of some reasons - could you document them in comment? Best regards, Krzysztof

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Fri, 15 Feb 2019 at 15:48, Sylwester Nawrocki <s.nawrocki(a)samsung.com> wrote: > > On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio > interfaces" and the driver will try to register there multiple times > same platform device for the secondary FIFO, which of course fails > miserably. To fix this we derive the secondary platform device name > from the primary device name. The secondary device name will now > be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec". > > The fixed platform_device_id table entry is removed as the secondary > device name is now dynamic and device/driver matching is done through > driver_override. > > Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Suggested-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > --- > sound/soc/samsung/i2s.c | 49 +++++++++++++++++++++++++------------- > sound/soc/samsung/odroid.c | 2 +- > 2 files changed, 33 insertions(+), 18 deletions(-) > > diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c > index e36c44e2f1bb..4a6dd86459bc 100644 > --- a/sound/soc/samsung/i2s.c > +++ b/sound/soc/samsung/i2s.c > @@ -1339,20 +1339,34 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) > /* Create platform device for the secondary PCM */ > static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) > { > - struct platform_device *pdev; > + struct platform_device *pdev_sec; > + const char *devname; > int ret; > > - pdev = platform_device_register_simple("samsung-i2s-sec", -1, NULL, 0); > - if (!pdev) > + devname = devm_kasprintf(&priv->pdev->dev, GFP_KERNEL, "%s-sec", > + dev_name(&priv->pdev->dev)); > + if (!devname) > return -ENOMEM; > > - ret = device_attach(&pdev->dev); > + pdev_sec = platform_device_alloc(devname, -1); > + if (!pdev_sec) > + return -ENOMEM; > + > + pdev_sec->driver_override = "samsung-i2s"; > + > + ret = platform_device_add(pdev_sec); > if (ret < 0) { > - dev_info(&pdev->dev, "device_attach() failed\n"); > + platform_device_put(pdev_sec); > return ret; > } > > - priv->pdev_sec = pdev; > + priv->pdev_sec = pdev_sec; > + > + ret = device_attach(&pdev_sec->dev); > + if (ret < 0) { > + dev_info(&pdev_sec->dev, "device_attach() failed\n"); Don't you need here platform_device_unregister()? Best regards, Krzysztof

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Fri, 15 Feb 2019 at 15:48, Sylwester Nawrocki <s.nawrocki(a)samsung.com> wrote: > > On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio > interfaces" and the driver will try to register there multiple times > same platform device for the secondary FIFO, which of course fails > miserably. To fix this we derive the secondary platform device name > from the primary device name. The secondary device name will now > be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec". > > The fixed platform_device_id table entry is removed as the secondary > device name is now dynamic and device/driver matching is done through > driver_override. > > Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Suggested-by: Marek Szyprowski <m.szyprowski(a)samsung.com> > Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > --- > sound/soc/samsung/i2s.c | 49 +++++++++++++++++++++++++------------- > sound/soc/samsung/odroid.c | 2 +- > 2 files changed, 33 insertions(+), 18 deletions(-) > > diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c > index e36c44e2f1bb..4a6dd86459bc 100644 > --- a/sound/soc/samsung/i2s.c > +++ b/sound/soc/samsung/i2s.c > @@ -1339,20 +1339,34 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) > /* Create platform device for the secondary PCM */ > static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) > { > - struct platform_device *pdev; > + struct platform_device *pdev_sec; > + const char *devname; > int ret; > > - pdev = platform_device_register_simple("samsung-i2s-sec", -1, NULL, 0); > - if (!pdev) > + devname = devm_kasprintf(&priv->pdev->dev, GFP_KERNEL, "%s-sec", > + dev_name(&priv->pdev->dev)); > + if (!devname) > return -ENOMEM; > > - ret = device_attach(&pdev->dev); > + pdev_sec = platform_device_alloc(devname, -1); > + if (!pdev_sec) > + return -ENOMEM; > + > + pdev_sec->driver_override = "samsung-i2s"; This is wrong, see: https://patchwork.kernel.org/project/linux-samsung-soc/list/?series=81639&s… Although there is ongoing discussion whether the drivers should use driver_override at all: https://www.mail-archive.com/linux-kernel@vger.kernel.org/msg1934411.html Best regards, Krzysztof

1 0

Re: [alsa-devel] [PATCH -next] ASoC: samsung: i2s: Fix return value check in i2s_create_secondary_device()
by Sylwester Nawrocki 18 Feb '19

18 Feb '19

On 2/16/19 02:20, Wei Yongjun wrote: > In case of error, the function platform_device_register_simple() returns > ERR_PTR() and never returns NULL. The NULL test in the return value > check should be replaced with IS_ERR(). Thanks for the patch, it looks good but I have already prepared other fix that removes the platform_device_register_simple() call entirely: "ASoC: samsung: i2s: Fix multiple "IIS multi" devices initialization" https://www.spinics.net/lists/kernel/msg3045092.html -- Thanks, Sylwester

1 0

Re: [alsa-devel] [PATCH -next] ASoC: wm8741: Make function 'wm8741_mute' static
by Charles Keepax 18 Feb '19

18 Feb '19

On Sat, Feb 16, 2019 at 01:35:56AM +0000, Wei Yongjun wrote: > Fixes the following sparse warning: > > sound/soc/codecs/wm8741.c:371:5: warning: > symbol 'wm8741_mute' was not declared. Should it be static? > > Fixes: 36b1599340b5 ("ASoC: wm8741: Add digital mute callback") > Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> > --- Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Thanks, Charles

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Expose sample resolution through proc interface
by Jussi Laako 18 Feb '19

18 Feb '19

At least some USB devices use (MSB-aligned) audio format larger than the actual resolution of the device. In order to expose the actual device resolution (bBitResolution), add extra field to the procfs stream info interface. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/card.h | 1 + sound/usb/format.c | 2 ++ sound/usb/proc.c | 1 + 3 files changed, 4 insertions(+) diff --git a/sound/usb/card.h b/sound/usb/card.h index ac785d15ced4..d9bb2c4a3bd9 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -14,6 +14,7 @@ struct audioformat { u64 formats; /* ALSA format bits */ unsigned int channels; /* # channels */ unsigned int fmt_type; /* USB audio format type (1-3) */ + unsigned int fmt_bits; /* number of significant bits */ unsigned int frame_size; /* samples per frame for non-audio */ int iface; /* interface number */ unsigned char altsetting; /* corresponding alternate setting */ diff --git a/sound/usb/format.c b/sound/usb/format.c index fd13ac11b136..d5b7cbcc2956 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -87,6 +87,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, } } + fp->fmt_bits = (unsigned int) sample_width; + if ((pcm_formats == 0) && (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED))) { /* some devices don't define this correctly... */ diff --git a/sound/usb/proc.c b/sound/usb/proc.c index e80c9d0749c9..ef9190530fd2 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -109,6 +109,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s if (subs->speed != USB_SPEED_FULL) snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); + snd_iprintf(buffer, " Bits: %d\n", fp->fmt_bits); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } -- 2.17.1

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Re: [alsa-devel] Multiple codecs on one sound card for multi-channel sound card
by Yann CARDAILLAC 18 Feb '19

18 Feb '19

Hi guys, Where are we now with TDM on several codecs on the same TDM channel ? I'm sorry to dig out this old thread but I couldn't really find any proper information about that. I only plan to use 4 channel TDM with two codecs. I'm using IMX6's SSI as master for both codec. What I've got working today is 1 codec with ssi as I2S master. Do you have any pointer on what is achieved so far and how to implement it ? I'm using kernel *v4.14.78*. Any pointer to DTS achieving that or audio card doing that kind of thing would already be an amazing first hint.! Regards, Yann >Got it. Thanks Arnaud. >-Caleb > >On Thu, Sep 17, 2015 at 9:09 AM, arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel> ><arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> wrote: >>>>>*> Le 17/09/2015 17:34, Caleb Crome a écrit : *>>>>>>* On Thu, Sep 17, 2015 at 7:33 AM, arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel> *>>>* <arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> wrote: *>>>>>>>>* hello, *>>>>>>>*> Le 17/09/2015 15:38, Caleb Crome a écrit : *>>>>>>>>>>* On Thu, Sep 17, 2015 at 1:51 AM, arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel> *>>>>>* <arnaud.mouiche at invoxia.com <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>> wrote: *>>>>>>>>>>>>* Hello Caleb, *>>>>>>>>>>>>* freescale SSI is fine with more than 2 channels. *>>>>>>* I have planed to publish a set of patch in this direction but we still *>>>>>>* have *>>>>>>* some corner cases to fix first (rare issues with channels alignment) *>>>>>>>>>>* I'd love to see anything you're willing to show :-) *>>>>>>>>>>>* Yet, this following is far enough to have it working. *>>>>>>* And since the max/min rate is at the end the intersection of what the *>>>>>>* SSI, *>>>>>>* the codec and the card are declaring, it will not change anything until *>>>>>>* you *>>>>>>* connect the SSI to a codec with more than 2 channels, or a multi-codec *>>>>>>* solution. *>>>>*> *>>>>* What if I change channels_max to 16 for capture and playback? Last *>>>>>* time I tried, it would not keep the channels synchronized on the TDM *>>>>>* bus. I tried playing with the FIFO settings, which helped, but it *>>>>>* still would not start each channel in the right slot reliably. *>>>>>* Perhaps I was using an older kernel. *>>>>>>>>* you put the finger on what's hurting ! ;) *>>>>>>>>* In fact, we made 8 channels working in reliable way on imx50, with *>>>>* predictive latency ... with an old kernel (3.17) ... based on freescale *>>>>* BSP... on sound/soc/imx/imx-ssi.c (something that doesn't exist outside *>>>>* freescale BSP) *>>>>>>>>* Well, just to say it is doable on any recent imx since the hardware is *>>>>* not *>>>>* the issue. But the job was a set of awful hacks. *>>>>>>* Sorry, I don't quite understand. Are you saying that on a recent *>>>* kernel (say 4.1 or 4.2), you think it will just work. Or are yo *>>>* saying that even on a modern kernel it's a set of awful hacks? And *>>>* should I use fsl_ssi.c or imx-ssi.c (device tree/non device tree) *>>>* version? *>>>>* - I'm saying it is working on recent upstream kernel, using *>>* sound/soc/fsl/fsl_ssi.c, but with *>>* some channel sync issues in rare conditions (at least 4 channels case, *>>* opening multiple SSI at the same time). *>>* So, you should test on your side with you setup, and simply be prepared to *>>* not see it working as a charm. *>>>>* - we made it work perfectly, but on an old freescale BSP, where freescale *>>* rewrites a SSI driver different from the already 2 drivers (device tree/non *>>* device tree) available today in upstream + lot of awful hacks => so, this is *>>* not a hardware issue. *>>>>* - when I will have time, I will start from the upstream driver, and make it *>>* work correctly with 8 channels (or more) *>>>>* arnaud *>>>>>>>>* In short, what's your best current recommendation to get it working today? *>>>>>>>* My plan was to re-do all this job on an upstream kernel, in a way it can *>>>>* be *>>>>* accepted. *>>>>* Unfortunately you will have to wait, I'm under pressure for other stuff *>>>>* right now. *>>>>>>* Heh, understood. If there's anything I can do to help, let me know. *>>>>>>* Thanks, *>>>* -Caleb *>>>>>>>* Regards, *>>>>* arnaud *>>>>>>>>>>>>>* I'll give it a try again. Like I said, we'd love to get our *>>>>>* 16-channel board up and running on the MX6, so I'm definitely *>>>>>* motivated to put some work in to get it reliable. *>>>>>>>>>>* Thanks, *>>>>>* -Caleb *>>>>>>> -- [image: SMILE] <http://www.smile.eu/> 20 rue des Jardins 92600 Asnières-sur-Seine *Yann CARDAILLAC* Ingénieur Linux Embarqué [image: email] yann.cardaillac(a)smile.fr [image: url] http://www.smile.eu [image: Twitter] <https://twitter.com/GroupeSmile> [image: Facebook] <https://www.facebook.com/smileopensource> [image: LinkedIn] <https://www.linkedin.com/company/smile> [image: Github] <https://github.com/Smile-SA> [image: eco] Pour la planète, n'imprimez ce mail que si c'est nécessaire

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[alsa-devel] [PATCH V3] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by S.j. Wang 18 Feb '19

18 Feb '19

The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v3 - Add Cc: <stable(a)vger.kernel.org> sound/soc/fsl/fsl_esai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 57b484768a58..afe67c865330 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); -- 1.9.1

1 0

[alsa-devel] [PATCH V2] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by S.j. Wang 18 Feb '19

18 Feb '19

Fixes commit 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to wrong register, correct it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v2 - add Fixes tag and cc stable kernel. sound/soc/fsl/fsl_esai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 57b484768a58..afe67c865330 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); -- 1.9.1

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Re: [alsa-devel] [PATCH -next] ASoC: samsung: i2s: Fix return value check in i2s_create_secondary_device()
by Krzysztof Kozlowski 18 Feb '19

18 Feb '19

On Sat, 16 Feb 2019 at 02:07, Wei Yongjun <weiyongjun1(a)huawei.com> wrote: > > In case of error, the function platform_device_register_simple() returns > ERR_PTR() and never returns NULL. The NULL test in the return value > check should be replaced with IS_ERR(). > > Fixes: 7196c64c7d0c ("ASoC: samsung: i2s: Restore support for the secondary PCM") > Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> > --- > sound/soc/samsung/i2s.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) Reviewed-by: Krzysztof Kozlowski <krzk(a)kernel.org> Best regards, Krzysztof

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Re: [alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Daniel Baluta 18 Feb '19

18 Feb '19

On Fri, Feb 15, 2019 at 8:43 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Fri, 15 Feb 2019 18:31:08 +0100, > Mark Brown wrote: > > > > On Fri, Feb 15, 2019 at 06:21:04PM +0100, Takashi Iwai wrote: > > > > > Actually it's not all. As mentioned in my earlier post, there are > > > more of refcount unbalance in the code. Lots of them, really. > > > > > Below is a patch I've fixed quickly over the whole scanning the sound > > > tree. Some might be incorrect, so more detailed review for each piece > > > of change is needed... > > > > Extra references aren't such a problem for practical systems since we > > don't actually ever free any of the DT (there's no mainline way to use > > overlays yet...), that's how none of this stuff ever gets noticed. I'm > > not surprised there's loads of missing frees TBH. In contrast if we've > > got extra frees they will get noticed and cause problems. > > Yeah, the rest are just leaks, and we have already tons of them :) > > The point is that the commit in question introduced multiple such > issues, so they should be addressed as well. > > But, the biggest problem is that OF api is very hard to use right... Hello Mark/Takashi, Just sent this https://lkml.org/lkml/2019/2/16/52 to fix the problem that was bugging me and to update the patch from Kuninori. Feel free to send a patch for the rest of the refcount issues :). Kuninori, if you consider necessary please add your Signed-off-by to the patch as it is mostly your work but I didn't know exactly how to give you credit for it :(. thanks, Daniel.

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Re: [alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Kuninori Morimoto 18 Feb '19

18 Feb '19

Hi Takashi-san > > commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card") > > merged simple-card and simple-scu-card. Then it had refcount > > underflow bug. This patch fixup it. > > We will get below error without this patch. > > > > OF: ERROR: Bad of_node_put() on /sound > > CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514 > > Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) > > Workqueue: events deferred_probe_work_func > > Call trace: > > dump_backtrace+0x0/0x150 > > show_stack+0x24/0x30 > > dump_stack+0xb0/0xec > > of_node_release+0xd0/0xd8 > > kobject_put+0x74/0xe8 > > of_node_put+0x24/0x30 > > __of_get_next_child+0x50/0x70 > > of_get_next_child+0x40/0x68 > > asoc_simple_card_probe+0x604/0x730 > > platform_drv_probe+0x58/0xa8 > > ... > > Reported-by: Vicente Bergas <vicencb(a)gmail.com> > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > Please don't forget to put Fixes tag for a regression fix. Oops, indeed. Thanks. will add it on v2 patch. > And, looking at the code in 5.0-rc, it seems still leaving the > reference in some condition. The 5.0-rc code corrected your patch > looks like: > > node = of_get_child_by_name(top, PREFIX "dai-link"); > if (!node) { > node = of_node_get(top); > loop = 0; > } > > do { > .... > node = of_get_next_child(top, node); > } while (loop && node); > > So when loop=0, it aborts the loop at the first iteration. If node is > non-NULL at that point, it still keeps the refcount taken in > of_get_next_child(). > > That is, you'd need to call of_node_put(node) after the loop. Ah.. yes.. > And, this leads me checking the whole code, and I'm afraid that we > have many similar unblanced refcounts. For example, a tricky one is > for_each_child_of_node() loop. When you abort at the middle of the > loop, you'd need to unreference the remaining node. But we return or > abort immediately without unreferencing in many places. > > Hrm... Yes, I had noticed this refcounts issue. The code can be very confusable for this refcounts... Best regards --- Kuninori Morimoto

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Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by Fabio Estevam 16 Feb '19

16 Feb '19

Hi Shengjiu, On Fri, Feb 15, 2019 at 9:04 AM S.j. Wang <shengjiu.wang(a)nxp.com> wrote: > > The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to > wrong register, correct it. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Please add a Fixes tag and Cc stable. Also, it seems that Mark Brown is not on Cc. Please Cc him in v2. Thanks

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Re: [alsa-devel] Saffire Pro 26 IO Restarting Issue
by Takashi Sakamoto 15 Feb '19

15 Feb '19

Hi, On Fri, Feb 15, 2019 at 12:49:19PM +0000, ackolonges fds wrote: > Hi All, > > I was wondering if there might be anyone on here that could give me any > pointers in getting my Focusrite Saffire Pro 26 i/o interface working with > Linux via FFADO? > > It's a new desktop PC (Ryzen 5 2600, 8GB RAM), dual booting Windows 10 and > Ubuntu 18.10, with a PCI Express, Texas Instruments XIO2213 Firewire card. > > I have tested this Saffire interface successfully on a MacBook Pro (it was > plug and play), and also on this very desktop computer in Windows 10. > However, when the interface is connected in Ubuntu 18.10, it repeatedly > and periodically resets itself about 10-15 seconds after booting up, in a > loop. If I run ffado-dbus-server while it's booting, it connects > successfully to the interface, and quickly running FFADO mixer afterwards > lets me see the controls available, for a short time until the interface > resets itself, cutting all the connections. > > Has anybody here ever heard of this kind of issue before? Or perhaps > someone may have some ideas that might aid in troubleshooting the issue? > > Thanks so much for any advice you may have, In case of Focusrite Saffire Pro I/O 10 and 26, a combination of ALSA bebob driver and pulseaudio brings this problem due to quirk of these two models reported in bugzilla.kernel.org[1]. I've already committed a solution and it was merged to pulseaudio[2], however it has never included into any releases of pulseaudio at present. Therefore you can see this issue on any desktop environment in released distribution such as the latest Ubuntu. You can avoid this issue by two handy solutions at a quick glance: 1. suppress pulseaudio to handle your device 2. suppress your system to load ALSA bebob driver For solution 1, it's the best way to edit 90-pulseaudio.rules in your system. $ systemctl --user stop pulseaudio.service pulseaudio.socket $ sudo nano /lib/udev/rules.d/90-pulseaudio.rules (modify the above file according to my patch[2].) $ systemctl --user start pulseaudio.service pulseaudio.socket For solution 2, it's the easiest way to add ALSA bebob driver to blacklist. $ sudo nano /etc/modprobe.d/blacklist-bebob.conf (add entry of 'blacklist snd-bebob') (then unplug and plugin your device) Sorry for the inconvenience. [1] https://bugzilla.kernel.org/show_bug.cgi?id=199365 [2] https://gitlab.freedesktop.org/pulseaudio/pulseaudio/commit/37358e42c49abe3… Regards Takashi Sakamoto

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Re: [alsa-devel] [PATCH v5 0/3] Add NXP AUDMIX device and machine drivers
by Nicolin Chen 15 Feb '19

15 Feb '19

On Fri, Feb 15, 2019 at 02:01:32PM +0000, Viorel Suman wrote: > The patchset adds NXP Audio Mixer (AUDMIX) device and machine > drivers and related DT bindings documentation. For this series, Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> And Rob gave his at the previous version already. Thanks. > Changes since V4: > 1. Removed "model" attribute from device driver DT bindings documentation > as suggested by Nicolin. > > 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 | 50 ++ > sound/soc/fsl/Kconfig | 16 + > sound/soc/fsl/Makefile | 5 + > sound/soc/fsl/fsl_audmix.c | 578 +++++++++++++++++++++ > sound/soc/fsl/fsl_audmix.h | 102 ++++ > sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ > 6 files changed, 1078 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 >

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Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix register setting issue in RIGHT_J mode
by Nicolin Chen 15 Feb '19

15 Feb '19

On Fri, Feb 15, 2019 at 11:04:38AM +0000, S.j. Wang wrote: > The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to > wrong register, correct it. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Would need this for stable kernel too. Ackedy-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Thanks. > --- > sound/soc/fsl/fsl_esai.c | 7 ++++--- > 1 file changed, 4 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c > index 57b484768a58..afe67c865330 100644 > --- a/sound/soc/fsl/fsl_esai.c > +++ b/sound/soc/fsl/fsl_esai.c > @@ -398,7 +398,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) > break; > case SND_SOC_DAIFMT_RIGHT_J: > /* Data on rising edge of bclk, frame high, right aligned */ > - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; > + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; > + xcr |= ESAI_xCR_xWA; > break; > case SND_SOC_DAIFMT_DSP_A: > /* Data on rising edge of bclk, frame high, 1clk before data */ > @@ -455,12 +456,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) > return -EINVAL; > } > > - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; > + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; > regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); > regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); > > mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | > - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; > + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; > regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); > regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); > > -- > 1.9.1 >

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[alsa-devel] Applied "ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag" to the asoc tree
by Mark Brown 15 Feb '19

15 Feb '19

The patch ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag 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 76d9c68b360f852e784170f10cb431e4713c7d0b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 16:45:55 +0100 Subject: [PATCH] ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag There is now no users of this flag so remove it together with related code. The chan_name field of snd_dmaengine_dai_dma_data data structure is not removed as it is still in use by the PXA platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/dmaengine_pcm.h | 4 ---- sound/soc/soc-generic-dmaengine-pcm.c | 21 ++++----------------- 2 files changed, 4 insertions(+), 21 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 2c4cfaa135a6..c679f6116580 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -99,10 +99,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 1b44e363c50c..f1ab6285a085 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -265,7 +265,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_component_to_pcm(component); const struct snd_dmaengine_pcm_config *config = pcm->config; struct device *dev = component->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -285,19 +284,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - ((pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME) || - (config && config->chan_names[i]))) { - const char *chan_name = dma_data->chan_name; - - if (config && config->chan_names[i]) - chan_name = config->chan_names[i]; - + if (!pcm->chan[i] && config && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, - chan_name); - } + config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, @@ -420,10 +409,8 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, const char *name; struct dma_chan *chan; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - (!dev->of_node && !(config && config->dma_dev && - config->dma_dev->of_node))) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && + !(config && config->dma_dev && config->dma_dev->of_node))) return 0; if (config && config->dma_dev) { -- 2.20.1

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Prevent potential NULL platform data dereference
by Sylwester Nawrocki 15 Feb '19

15 Feb '19

When np is NULL i2s_pdata could also be NULL but i2s_pdata is now being dereferenced without proper check. Fix this and shorten the error message so we don't exceed 80 characters limit. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03fff1c657be..02472f576e17 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1369,7 +1369,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) struct i2s_dai *pri_dai, *sec_dai = NULL; struct s3c_audio_pdata *i2s_pdata = pdev->dev.platform_data; struct resource *res; - u32 regs_base, quirks = 0, idma_addr = 0; + u32 regs_base, idma_addr = 0; struct device_node *np = pdev->dev.of_node; const struct samsung_i2s_dai_data *i2s_dai_data; int num_dais, ret; @@ -1389,11 +1389,19 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - quirks = np ? i2s_dai_data->quirks : i2s_pdata->type.quirks; - num_dais = (quirks & QUIRK_SEC_DAI) ? 2 : 1; + if (np) { + priv->quirks = i2s_dai_data->quirks; + } else { + if (!i2s_pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -EINVAL; + } + priv->quirks = i2s_pdata->type.quirks; + } + + num_dais = (priv->quirks & QUIRK_SEC_DAI) ? 2 : 1; priv->pdev = pdev; priv->variant_regs = i2s_dai_data->i2s_variant_regs; - priv->quirks = quirks; ret = i2s_alloc_dais(priv, i2s_dai_data, num_dais); if (ret < 0) @@ -1405,11 +1413,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) spin_lock_init(&priv->pcm_lock); if (!np) { - if (i2s_pdata == NULL) { - dev_err(&pdev->dev, "Can't work without s3c_audio_pdata\n"); - return -EINVAL; - } - pri_dai->dma_playback.filter_data = i2s_pdata->dma_playback; pri_dai->dma_capture.filter_data = i2s_pdata->dma_capture; pri_dai->filter = i2s_pdata->dma_filter; @@ -1418,7 +1421,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } else { if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { - if (quirks & QUIRK_SUPPORTS_IDMA) { + if (priv->quirks & QUIRK_SUPPORTS_IDMA) { dev_info(&pdev->dev, "idma address is not"\ "specified"); } @@ -1451,7 +1454,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.addr_width = 4; pri_dai->priv = priv; - if (quirks & QUIRK_PRI_6CHAN) + if (priv->quirks & QUIRK_PRI_6CHAN) pri_dai->drv->playback.channels_max = 6; ret = samsung_asoc_dma_platform_register(&pdev->dev, pri_dai->filter, @@ -1459,7 +1462,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_disable_clk; - if (quirks & QUIRK_SEC_DAI) { + if (priv->quirks & QUIRK_SEC_DAI) { sec_dai = &priv->dai[SAMSUNG_I2S_ID_SECONDARY - 1]; sec_dai->dma_playback.addr = regs_base + I2STXDS; -- 2.17.1

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Re: [alsa-devel] [PATCH] ASoC: simple-card: fixup refcount_t underflow
by Mark Brown 15 Feb '19

15 Feb '19

On Fri, Feb 15, 2019 at 06:21:04PM +0100, Takashi Iwai wrote: > Actually it's not all. As mentioned in my earlier post, there are > more of refcount unbalance in the code. Lots of them, really. > Below is a patch I've fixed quickly over the whole scanning the sound > tree. Some might be incorrect, so more detailed review for each piece > of change is needed... Extra references aren't such a problem for practical systems since we don't actually ever free any of the DT (there's no mainline way to use overlays yet...), that's how none of this stuff ever gets noticed. I'm not surprised there's loads of missing frees TBH. In contrast if we've got extra frees they will get noticed and cause problems.

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[alsa-devel] [regression] snd_soc_simple_card: refcount_t: underflow; use-after-free.
by Vicente Bergas 15 Feb '19

15 Feb '19

Hi, in linux 5.0-rc there is a regression regarding snd_soc_simple_card. Since updated from 4.20 there is a new error appearing in the kernel log, although sound works fine and the system is stable. The issue has not been bisected, but it probably lies at or arround this commit: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… The DT node referred in the trace below is: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… The kernel comes from the ArchLinuxArm distribution, its configuration is: https://archlinuxarm.org/packages/aarch64/linux-aarch64-rc/files/config This trace is from "dmesg -tl1,2,3,4": OF: ERROR: Bad of_node_put() on /hdmi-sound CPU: 2 PID: 370 Comm: kworker/2:2 Tainted: G C 5.0.0-rc6-1-ARCH #1 Hardware name: Sapphire-RK3399 Board (DT) Workqueue: events deferred_probe_work_func Call trace: dump_backtrace+0x0/0x1b8 show_stack+0x24/0x30 dump_stack+0x98/0xbc of_node_release+0xd0/0xd8 kobject_put+0x8c/0x1f0 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x64/0x90 asoc_simple_card_probe+0xe4/0x6b0 [snd_soc_simple_card] platform_drv_probe+0x58/0xa8 really_probe+0x1f0/0x3d8 driver_probe_device+0xe4/0x138 __device_attach_driver+0xb4/0x140 bus_for_each_drv+0x8c/0xd8 __device_attach+0xdc/0x158 device_initial_probe+0x24/0x30 bus_probe_device+0x9c/0xa8 deferred_probe_work_func+0xa0/0xf0 process_one_work+0x1ac/0x400 worker_thread+0x50/0x488 kthread+0x130/0x138 ret_from_fork+0x10/0x1c ------------[ cut here ]------------ refcount_t: underflow; use-after-free. WARNING: CPU: 2 PID: 370 at lib/refcount.c:187 refcount_sub_and_test_checked+0xb8/0xd0 Modules linked in: snd_soc_hdmi_codec rockchip_vpu(C+) rockchip_rga videobuf2_dma_contig videobuf2_dma_sg v4l2_mem2mem videobuf2_vmalloc videobuf2_memops videobuf2_v4l2 videobuf2_common rc_cec snd_soc_simple_card realtek snd_soc_rockchip_i2s snd_soc_simple_card_utils snd_soc_rockchip_pcm dw_hdmi_cec dw_hdmi_i2s_audio dw_wdt videodev rtc_rk808 media hid_kensington dwmac_rk rockchip_saradc rockchip_thermal stmmac_platform stmmac squashfs loop crypto_user gpio_keys rockchipdrm analogix_dp dw_hdmi cec rc_core dw_mipi_dsi drm_kms_helper syscopyarea sysfillrect sysimgblt fb_sys_fops drm drm_panel_orientation_quirks CPU: 2 PID: 370 Comm: kworker/2:2 Tainted: G C 5.0.0-rc6-1-ARCH #1 Hardware name: Sapphire-RK3399 Board (DT) Workqueue: events deferred_probe_work_func pstate: 80000085 (Nzcv daIf -PAN -UAO) pc : refcount_sub_and_test_checked+0xb8/0xd0 lr : refcount_sub_and_test_checked+0xb8/0xd0 sp : ffff000012d9ba20 x29: ffff000012d9ba20 x28: 0000000000000000 x27: 0000000000000002 x26: 0000000000000001 x25: ffff0000115ad6c8 x24: ffff0000090bb428 x23: ffff8000f781a740 x22: 0000000000000000 x21: ffff8000f781a740 x20: ffff8000f781a740 x19: ffff8000f781a790 x18: 0000000000000000 x17: 0000000000000000 x16: 0000000000000000 x15: ffffffffffffffff x14: 0000000000000003 x13: 0000000000000000 x12: ffff000011810000 x11: ffff0000115d6000 x10: ffff000011810f48 x9 : 0000000000000000 x8 : ffff00001181f170 x7 : 0000000000000000 x6 : 0000000000000001 x5 : 0000000000000000 x4 : 0000000000000001 x3 : 0000000000000007 x2 : 0000000000000007 x1 : 86b2dbbfc7425b00 x0 : 0000000000000000 Call trace: refcount_sub_and_test_checked+0xb8/0xd0 refcount_dec_and_test_checked+0x14/0x20 kobject_put+0x24/0x1f0 of_node_put+0x24/0x30 __of_get_next_child+0x50/0x70 of_get_next_child+0x64/0x90 asoc_simple_card_probe+0x544/0x6b0 [snd_soc_simple_card] platform_drv_probe+0x58/0xa8 really_probe+0x1f0/0x3d8 driver_probe_device+0xe4/0x138 __device_attach_driver+0xb4/0x140 bus_for_each_drv+0x8c/0xd8 __device_attach+0xdc/0x158 device_initial_probe+0x24/0x30 bus_probe_device+0x9c/0xa8 deferred_probe_work_func+0xa0/0xf0 process_one_work+0x1ac/0x400 worker_thread+0x50/0x488 kthread+0x130/0x138 ret_from_fork+0x10/0x1c ---[ end trace ae290e9394a14a2f ]--- asoc-simple-card hdmi-sound: ASoC: no DMI vendor name! Regards, Vicente.

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[alsa-devel] [PATCH v4] ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag
by Sylwester Nawrocki 14 Feb '19

14 Feb '19

There is now no users of this flag so remove it together with related code. The chan_name field of snd_dmaengine_dai_dma_data data structure is not removed as it is still in use by the PXA platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Changes since v3: - dropped removal of chan_name field. include/sound/dmaengine_pcm.h | 4 ---- sound/soc/soc-generic-dmaengine-pcm.c | 21 ++++----------------- 2 files changed, 4 insertions(+), 21 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 2c4cfaa135a6..c679f6116580 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -99,10 +99,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 1b44e363c50c..f1ab6285a085 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -265,7 +265,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_component_to_pcm(component); const struct snd_dmaengine_pcm_config *config = pcm->config; struct device *dev = component->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -285,19 +284,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - ((pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME) || - (config && config->chan_names[i]))) { - const char *chan_name = dma_data->chan_name; - - if (config && config->chan_names[i]) - chan_name = config->chan_names[i]; - + if (!pcm->chan[i] && config && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, - chan_name); - } + config->chan_names[i]); if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, @@ -420,10 +409,8 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, const char *name; struct dma_chan *chan; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - (!dev->of_node && !(config && config->dma_dev && - config->dma_dev->of_node))) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && + !(config && config->dma_dev && config->dma_dev->of_node))) return 0; if (config && config->dma_dev) { -- 2.17.1

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[alsa-devel] Applied "ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage 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 eb540d3988d93d2b0231a5b36012aa0b3abaec81 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:44 +0100 Subject: [PATCH] ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage If the probe call is on the primary DAI we can use 'other' in place of i2s->sec_dai, if the probe call is on the secondary DAI we can use 'i2s' in place of other->sec_dai. While at it fix one whitespace issue. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 6ab99e38e6dd..967c1b22ac35 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1057,18 +1057,17 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(dai->dev); if (is_secondary(i2s)) { /* If this is probe on the secondary DAI */ - snd_soc_dai_init_dma_data(dai, &other->sec_dai->dma_playback, - NULL); + snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, NULL); } else { snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, - &i2s->dma_capture); + &i2s->dma_capture); if (priv->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, priv->addr + I2SCON); if (priv->quirks & QUIRK_SUPPORTS_IDMA) idma_reg_addr_init(priv->addr, - i2s->sec_dai->idma_playback.addr); + other->idma_playback.addr); } /* Reset any constraint on RFS and BFS */ -- 2.20.1

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[alsa-devel] Applied "ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition 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 c5ba619247391527248c4a8fb27e68f7cece8d0d Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:45 +0100 Subject: [PATCH] ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition Change indentation so this macro definition spans 2 rows and looks more consistent with surrounding code. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 967c1b22ac35..07f815a730df 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1150,9 +1150,8 @@ static const struct snd_soc_component_driver samsung_i2s_component = { .num_dapm_routes = ARRAY_SIZE(samsung_i2s_dapm_routes), }; -#define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE) +#define SAMSUNG_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static int i2s_alloc_dais(struct samsung_i2s_priv *priv, const struct samsung_i2s_dai_data *i2s_dai_data, -- 2.20.1

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[alsa-devel] Applied "ASoC: samsung: i2s: Comments clean up" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Comments clean up 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 9f9f8a5b79b0855d162153de41ffda687fd2241f Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:46 +0100 Subject: [PATCH] ASoC: samsung: i2s: Comments clean up Spelling error fixes, upper/lower case letter changes. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 29 +++++++++++++++-------------- 1 file changed, 15 insertions(+), 14 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 07f815a730df..84cfa2c0ba68 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1,5 +1,4 @@ -/* sound/soc/samsung/i2s.c - * +/* * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. @@ -61,10 +60,10 @@ struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* Frame Clock */ + /* Frame clock */ unsigned frmclk; /* - * Specifically requested RCLK,BCLK by MACHINE Driver. + * Specifically requested RCLK, BCLK by machine driver. * 0 indicates CPU driver is free to choose any value. */ unsigned rfs, bfs; @@ -72,8 +71,9 @@ struct i2s_dai { struct i2s_dai *pri_dai; /* Pointer to the Secondary_Fifo if it has one, NULL otherwise */ struct i2s_dai *sec_dai; -#define DAI_OPENED (1 << 0) /* Dai is opened */ -#define DAI_MANAGER (1 << 1) /* Dai is the manager */ + +#define DAI_OPENED (1 << 0) /* DAI is opened */ +#define DAI_MANAGER (1 << 1) /* DAI is the manager */ unsigned mode; /* Driver for this DAI */ @@ -98,7 +98,7 @@ struct samsung_i2s_priv { /* Spinlock protecting access to the device's registers */ spinlock_t lock; - /* Lock for cross i/f checks */ + /* Lock for cross interface checks */ spinlock_t pcm_lock; /* CPU DAIs and their corresponding drivers */ @@ -309,7 +309,7 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) writel(mod, priv->addr + I2SMOD); } -/* Read Bit-Clock of I2S (in multiples of LRCLK) */ +/* Read bit-clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { struct samsung_i2s_priv *priv = i2s->priv; @@ -331,7 +331,7 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) } } -/* Write Bit-Clock of I2S (in multiples of LRCLK) */ +/* Write bit-clock of I2S (in multiples of LRCLK) */ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { struct samsung_i2s_priv *priv = i2s->priv; @@ -383,7 +383,7 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) writel(mod, priv->addr + I2SMOD); } -/* Sample-Size */ +/* Sample size */ static inline int get_blc(struct i2s_dai *i2s) { int blc = readl(i2s->priv->addr + I2SMOD); @@ -397,7 +397,7 @@ static inline int get_blc(struct i2s_dai *i2s) } } -/* TX Channel Control */ +/* TX channel control */ static void i2s_txctrl(struct i2s_dai *i2s, int on) { struct samsung_i2s_priv *priv = i2s->priv; @@ -742,7 +742,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; - /* fall through */ + /* Fall through */ case 4: val |= MOD_DC1_EN; break; @@ -821,7 +821,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -/* We set constraints on the substream acc to the version of I2S */ +/* We set constraints on the substream according to the version of I2S */ static int i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1056,7 +1056,8 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(dai->dev); - if (is_secondary(i2s)) { /* If this is probe on the secondary DAI */ + if (is_secondary(i2s)) { + /* If this is probe on the secondary DAI */ snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, NULL); } else { snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, -- 2.20.1

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[alsa-devel] [PATCH v3 Resend 00/13] ASoC: dmaengine updates, secondary CPU DAI for Odroid
by Sylwester Nawrocki 14 Feb '19

14 Feb '19

This is a rebased version with the build break fixed and added Krzysztof's Acks. My apologies, I double checked there is no build errors between patches this time. Below is cover letter text from v2. This series restores support for the secondary CPU DAI in samsung/i2s.c driver and adds support for secondary CPU DAI on Odroid boards. I have added 4 more clean up patches comparing to first version of the patch set (16/22...19/22). The first patch adds new platform device for the secondary PCM, I tried to avoid registering new platform device in probe() as it doesn't play well with deferred probing but couldn't find a way to avoid that new device, unique devices are needed to register "PCM" components for each CPU DAI. Patches 2/21...11/21 are mostly a clean up, following conversion to single component with multiple DAIs, allowing to reference both primary/secondary DAI in DT. The patch set has been tested on Odroid U3, XU3, XU4 and Mini2440 (s3c2440 non-DT) boards. Changes since v1 ((addressing Krzysztof's review comments): - added a comment, whitespace change in patch 1/21 - dropped unnecessary new line addition from patch 1/21 Sylwester Nawrocki (13): ASoC: samsung: i2s: Move SFR pointer to common driver data structure ASoC: samsung: i2s: Drop spinlock pointer from i2s_dai data structure ASoC: samsung: i2s: Move IP variant data to common driver data structure ASoC: samsung: i2s: Move quirks data to common driver data structure ASoC: samsung: i2s: Get rid of a static spinlock ASoC: samsung: odroid: Add support for secondary CPU DAI ASoC: samsung: Specify DMA channel names through custom DMA config ASoC: samsung: Drop DAI DMA data chan_name assignments ASoC: dmaengine: Remove unused SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag ASoC: samsung: i2s: Simplify pri_dai, sec_dai pointers usage ASoC: samsung: i2s: Change indentation in SAMSUNG_I2S_FMTS definition ASoC: samsung: i2s: Comments clean up ASoC: samsung: i2s: Convert to SPDX License Indentifier include/sound/dmaengine_pcm.h | 6 - sound/soc/samsung/dmaengine.c | 12 +- sound/soc/samsung/i2s.c | 297 +++++++++++++------------- sound/soc/samsung/odroid.c | 131 ++++++++---- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 21 +- 7 files changed, 250 insertions(+), 225 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v2 0/2] ASoC: da7219: BCLK and TDM flexibility improvements
by Adam Thomson 14 Feb '19

14 Feb '19

This patch set updates the driver to be more flexible regarding the BCLK and TDM configurations of the device. Previously the BCLK rate was fixed at 64 periods per WCLK, when the codec is DAI clock master, but to satisfy devices which prefer a lower rate, and to save power, the BCLK rate is now calculated based on hw_params() data. As a fallout of the BCLK efforts it was also apparent that the TDM code could be made more flexible as well. The device is capable of automatically detecting the BCLK rate when it is clock slave, so previous limitations imposed were not necessary. The mask handling is now used as an input to determine the BCLK offset to align closer with other examples within ALSA. Changes in v2: - Update 0001 patch to use '&&' instead of '&' in hw_params() to decide if BCLK rate should be configured or not. Adam Thomson (2): ASoC: da7219: Add support for master mode BCLK rate adjustment ASoC: da7219: Update TDM usage to be more flexible sound/soc/codecs/da7219.c | 116 +++++++++++++++++++++++++++++----------------- sound/soc/codecs/da7219.h | 1 + 2 files changed, 74 insertions(+), 43 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: da7219: Update TDM usage to be more flexible" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: da7219: Update TDM usage to be more flexible 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 541ccdc113f000d51858ee7e135889e4096a3316 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Thu, 14 Feb 2019 10:13:30 +0000 Subject: [PATCH] ASoC: da7219: Update TDM usage to be more flexible The previous implementatation was restrictive with regards to BCLK rates for slave mode where the driver would not allow rates the codec couldn't provide itself as clock master. The codec is able to automatically determine and handle whatever rate is provided so this restriction isn't necessary for slave mode. The code was also flawed with regards to setting of the frame offset as using rx_mask to explicitly set the offset has the knock on effect of impacting the min and max channels for the codec, in soc_pcm_hw_params() through the call to soc_pcm_codec_params_fixup(). With this update, the driver now only limits frame size if codec is clock master, and dynamically determines the BCLK offset relating to WCLK using the tx_mask for slot offset along with the slot width provided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 80 +++++++++++++++++++++++---------------- 1 file changed, 47 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c599aa9f609b..121a8190f93e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1391,8 +1391,10 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, { struct snd_soc_component *component = dai->component; struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); - u8 dai_bclks_per_wclk; - __le16 offset; + unsigned int ch_mask; + u8 dai_bclks_per_wclk, slot_offset; + u16 offset; + __le16 dai_offset; u32 frame_size; /* No channels enabled so disable TDM */ @@ -1405,51 +1407,63 @@ static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai, } /* Check we have valid slots */ - if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) { - dev_err(component->dev, "Invalid number of slots, max = %d\n", + slot_offset = ffs(tx_mask) - 1; + ch_mask = (tx_mask >> slot_offset); + if (fls(ch_mask) > DA7219_DAI_TDM_MAX_SLOTS) { + dev_err(component->dev, + "Invalid number of slots, max = %d\n", DA7219_DAI_TDM_MAX_SLOTS); return -EINVAL; } - /* Check we have a valid offset given */ - if (rx_mask > DA7219_DAI_OFFSET_MAX) { - dev_err(component->dev, "Invalid slot offset, max = %d\n", - DA7219_DAI_OFFSET_MAX); + /* + * Ensure we have a valid offset into the frame, based on slot width + * and slot offset of first slot we're interested in. + */ + offset = slot_offset * slot_width; + if (offset > DA7219_DAI_OFFSET_MAX) { + dev_err(component->dev, "Invalid frame offset %d\n", offset); return -EINVAL; } - /* Calculate & validate frame size based on slot info provided. */ - frame_size = slots * slot_width; - switch (frame_size) { - case 32: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; - break; - case 64: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; - break; - case 128: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; - break; - case 256: - dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; - break; - default: - dev_err(component->dev, "Invalid frame size %d\n", frame_size); - return -EINVAL; - } + /* + * If we're master, calculate & validate frame size based on slot info + * provided as we have a limited set of rates available. + */ + if (da7219->master) { + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(component->dev, "Invalid frame size %d\n", + frame_size); + return -EINVAL; + } - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, - DA7219_DAI_BCLKS_PER_WCLK_MASK, - dai_bclks_per_wclk); + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + } - offset = cpu_to_le16(rx_mask); + dai_offset = cpu_to_le16(offset); regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, - &offset, sizeof(offset)); + &dai_offset, sizeof(dai_offset)); snd_soc_component_update_bits(component, DA7219_DAI_TDM_CTRL, DA7219_DAI_TDM_CH_EN_MASK | DA7219_DAI_TDM_MODE_EN_MASK, - (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | + (ch_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | DA7219_DAI_TDM_MODE_EN_MASK); da7219->tdm_en = true; -- 2.20.1

1 0

[alsa-devel] [PATCH v2 00/20] ASoC: Sound Open Firmware (SOF) - Intel support
by Pierre-Louis Bossart 14 Feb '19

14 Feb '19

This patch series introduces the support for Intel devices (Baytrail, CherryTrail, Broadwell/Haswell, ApolloLake, GeminiLake, CannonLake, IceLake). Support for Skylake and KabyLake is incomplete at the moment but the basic building blocks required are already present. The series starts with the 'legacy' devices, then all the more recent ones and concludes with ACPI/PCI hooks. SKL+ platforms can be compiled without HDaudio link/codec support, in which case the hdac library is not used. When support for HDMI or HDaudio codecs is selected, parts of this library are used. The machine drivers supported by SOF are not provided as part of this series. The build will only work in 'nocodec' mode for now. Support for SPI-based devices, compressed streams is also not provided for now since it's not ready for upstream just yet. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Changes since v2: Addressed dozens of comments from Takashi Iwai, Mark Brown, Andy Shevchenko, Daniel Baluta (Thanks!) Transition to new display power management Fixed suspend issues Optimized register polling, better handling of timeouts and firmware download retries Detect number of HDaudio codec and select machine driver accordingly Added workqueue based probe for HDaudio (dependency on modules) Fixed platform name override and conflicts with DT deferred probe changes Added prefix for firmware and topology files and module arguments Split Baytrail and Edison Fixed warnings (smatch, cppcheck) Keyon Jie (2): ASoC: SOF: Intel: Add hda-bus support and initialization ASoC: SOF: Intel: SKL, CNL, APL platform DAIs Liam Girdwood (15): ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support. ASoC: SOF: Intel: Add HSW HW DSP support ASoC: SOF: Intel: Add BDW HW DSP support ASoC: SOF: Intel: Add APL/CNL HW DSP support ASoC: SOF: Intel: Add HDA controller for Intel DSP ASoC: SOF: Intel: Add Intel specific HDA DSP HW operations ASoC: SOF: Intel: Add Intel specific HDA IPC mechanisms. ASoC: SOF: Intel: Add Intel specific HDA firmware loader ASoC: SOF: Intel: Add Intel specific HDA PCM operations ASoC: SOF: Intel: Add Intel specific HDA stream operations ASoC: SOF: Intel: Add Intel specific HDA trace operations ASoC: SOF: Intel: Add platform differentiation for SKL, APL and CNL ASoC: SOF: Add ACPI device support ASoC: SOF: Add PCI device support ASoC: SOF: Add Build support for SOF core and Intel drivers Pierre-Louis Bossart (3): ASoC: SOF: Intel: Add support for HDAudio codecs ASoC: SOF: Intel: Add SKL-specific code loader ASoC: Intel: Kconfig: expose common option between SST and SOF drivers sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/intel/Kconfig | 7 +- sound/soc/sof/Kconfig | 141 +++++ sound/soc/sof/Makefile | 18 + sound/soc/sof/intel/Kconfig | 253 ++++++++ sound/soc/sof/intel/Makefile | 17 + sound/soc/sof/intel/apl.c | 111 ++++ sound/soc/sof/intel/bdw.c | 716 +++++++++++++++++++++ sound/soc/sof/intel/byt.c | 893 +++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 269 ++++++++ sound/soc/sof/intel/hda-bus.c | 108 ++++ sound/soc/sof/intel/hda-codec.c | 171 +++++ sound/soc/sof/intel/hda-ctrl.c | 165 +++++ sound/soc/sof/intel/hda-dai.c | 362 +++++++++++ sound/soc/sof/intel/hda-dsp.c | 440 +++++++++++++ sound/soc/sof/intel/hda-ipc.c | 403 ++++++++++++ sound/soc/sof/intel/hda-loader-skl.c | 526 ++++++++++++++++ sound/soc/sof/intel/hda-loader.c | 359 +++++++++++ sound/soc/sof/intel/hda-pcm.c | 235 +++++++ sound/soc/sof/intel/hda-stream.c | 663 ++++++++++++++++++++ sound/soc/sof/intel/hda-trace.c | 98 +++ sound/soc/sof/intel/hda.c | 680 ++++++++++++++++++++ sound/soc/sof/intel/hda.h | 549 ++++++++++++++++ sound/soc/sof/intel/hsw.c | 717 +++++++++++++++++++++ sound/soc/sof/intel/shim.h | 177 ++++++ sound/soc/sof/intel/skl.c | 104 ++++ sound/soc/sof/sof-acpi-dev.c | 310 ++++++++++ sound/soc/sof/sof-pci-dev.c | 351 +++++++++++ 29 files changed, 8844 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/intel/Kconfig create mode 100644 sound/soc/sof/intel/Makefile create mode 100644 sound/soc/sof/intel/apl.c create mode 100644 sound/soc/sof/intel/bdw.c create mode 100644 sound/soc/sof/intel/byt.c create mode 100644 sound/soc/sof/intel/cnl.c create mode 100644 sound/soc/sof/intel/hda-bus.c create mode 100644 sound/soc/sof/intel/hda-codec.c create mode 100644 sound/soc/sof/intel/hda-ctrl.c create mode 100644 sound/soc/sof/intel/hda-dai.c create mode 100644 sound/soc/sof/intel/hda-dsp.c create mode 100644 sound/soc/sof/intel/hda-ipc.c create mode 100644 sound/soc/sof/intel/hda-loader-skl.c create mode 100644 sound/soc/sof/intel/hda-loader.c create mode 100644 sound/soc/sof/intel/hda-pcm.c create mode 100644 sound/soc/sof/intel/hda-stream.c create mode 100644 sound/soc/sof/intel/hda-trace.c create mode 100644 sound/soc/sof/intel/hda.c create mode 100644 sound/soc/sof/intel/hda.h create mode 100644 sound/soc/sof/intel/hsw.c create mode 100644 sound/soc/sof/intel/shim.h create mode 100644 sound/soc/sof/intel/skl.c create mode 100644 sound/soc/sof/sof-acpi-dev.c create mode 100644 sound/soc/sof/sof-pci-dev.c -- 2.17.1

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[alsa-devel] [PATCH] ASoC: stm32: sai: add missing put_device()
by Wen Yang 14 Feb '19

14 Feb '19

The of_find_device_by_node() takes a reference to the underlying device structure, we should release that reference. Fixes: 7dd0d835582f ("ASoC: stm32: sai: simplify sync modes management") Signed-off-by: Wen Yang <yellowriver2010(a)hotmail.com> --- sound/soc/stm/stm32_sai.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index bcb35ca..14c9591 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -112,16 +112,21 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!sai_provider) { dev_err(&sai_client->pdev->dev, "SAI sync provider data not found\n"); - return -EINVAL; + ret = -EINVAL; + goto out_put_dev; } /* Configure sync client */ ret = stm32_sai_sync_conf_client(sai_client, synci); if (ret < 0) - return ret; + goto out_put_dev; /* Configure sync provider */ - return stm32_sai_sync_conf_provider(sai_provider, synco); + ret = stm32_sai_sync_conf_provider(sai_provider, synco); + +out_put_dev: + put_device(&pdev->dev); + return ret; } static int stm32_sai_probe(struct platform_device *pdev) -- 2.7.4

3 4

[alsa-devel] Fwd: right way to enumerate pcm devices
by Joël Krähemann 14 Feb '19

14 Feb '19

---------- Forwarded message --------- From: Joël Krähemann <jkraehemann(a)gmail.com> Date: Thu, Feb 14, 2019 at 12:11 AM Subject: Re: [alsa-devel] right way to enumerate pcm devices To: <sylvain.bertrand(a)gmail.com> Hi all, Thank you for the hint about alsa hints. It was very helpful. Just updated my code: http://git.savannah.nongnu.org/cgit/gsequencer.git/tree/ags/audio/ags_devou… bests, Joël On Wed, Feb 13, 2019 at 11:44 PM <sylvain.bertrand(a)gmail.com> wrote: > > On Wed, Feb 13, 2019 at 10:25:18PM +0000, sylvain.bertrand(a)gmail.com wrote: > > ... > > Actually, I would just give a way to police the access to buffers. For > instance the user application would be in charge of handling the actual > mixing (I guess mmap-ed dma buffers). > > -- > Sylvain > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

[alsa-devel] Applied "ASoC: samsung: i2s: Move SFR pointer to common driver data structure" to the asoc tree
by Mark Brown 14 Feb '19

14 Feb '19

The patch ASoC: samsung: i2s: Move SFR pointer to common driver data structure 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 e2e16fa6a32dcf0a340cb2b6155a44d1bf5858ef Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Thu, 14 Feb 2019 10:37:35 +0100 Subject: [PATCH] ASoC: samsung: i2s: Move SFR pointer to common driver data structure The SFR region is common for both DAIs so move related data structure field from struct i2s_dai to the common driver data structure. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 106 ++++++++++++++++++++++------------------ 1 file changed, 59 insertions(+), 47 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 72f0cb7abb30..3f8955608a94 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -60,8 +60,7 @@ struct samsung_i2s_dai_data { struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* Memory mapped SFR region */ - void __iomem *addr; + /* Frame Clock */ unsigned frmclk; /* @@ -100,6 +99,9 @@ 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 spinlock; @@ -143,7 +145,9 @@ static inline bool is_secondary(struct i2s_dai *i2s) /* If operating in SoC-Slave mode */ static inline bool is_slave(struct i2s_dai *i2s) { - u32 mod = readl(i2s->addr + I2SMOD); + struct samsung_i2s_priv *priv = i2s->priv; + + u32 mod = readl(priv->addr + I2SMOD); return (mod & (1 << i2s->variant_regs->mss_off)) ? true : false; } @@ -155,7 +159,7 @@ static inline bool tx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SCON); + active = readl(i2s->priv->addr + I2SCON); if (is_secondary(i2s)) active &= CON_TXSDMA_ACTIVE; @@ -193,7 +197,7 @@ static inline bool rx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE; + active = readl(i2s->priv->addr + I2SCON) & CON_RXDMA_ACTIVE; return active ? true : false; } @@ -256,8 +260,10 @@ static inline bool is_manager(struct i2s_dai *i2s) /* Read RCLK of I2S (in multiples of LRCLK) */ static inline unsigned get_rfs(struct i2s_dai *i2s) { + struct samsung_i2s_priv *priv = i2s->priv; u32 rfs; - rfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->rfs_off; + + rfs = readl(priv->addr + I2SMOD) >> i2s->variant_regs->rfs_off; rfs &= i2s->variant_regs->rfs_mask; switch (rfs) { @@ -275,7 +281,8 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) /* Write RCLK of I2S (in multiples of LRCLK) */ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { - u32 mod = readl(i2s->addr + I2SMOD); + struct samsung_i2s_priv *priv = i2s->priv; + u32 mod = readl(priv->addr + I2SMOD); int rfs_shift = i2s->variant_regs->rfs_off; mod &= ~(i2s->variant_regs->rfs_mask << rfs_shift); @@ -307,14 +314,16 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) break; } - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); } /* Read Bit-Clock of I2S (in multiples of LRCLK) */ static inline unsigned get_bfs(struct i2s_dai *i2s) { + struct samsung_i2s_priv *priv = i2s->priv; u32 bfs; - bfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->bfs_off; + + bfs = readl(priv->addr + I2SMOD) >> i2s->variant_regs->bfs_off; bfs &= i2s->variant_regs->bfs_mask; switch (bfs) { @@ -333,7 +342,8 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) /* Write Bit-Clock of I2S (in multiples of LRCLK) */ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { - u32 mod = readl(i2s->addr + I2SMOD); + struct samsung_i2s_priv *priv = i2s->priv; + u32 mod = readl(priv->addr + I2SMOD); int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; int bfs_shift = i2s->variant_regs->bfs_off; @@ -378,13 +388,13 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) return; } - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); } /* Sample-Size */ static inline int get_blc(struct i2s_dai *i2s) { - int blc = readl(i2s->addr + I2SMOD); + int blc = readl(i2s->priv->addr + I2SMOD); blc = (blc >> 13) & 0x3; @@ -398,7 +408,8 @@ static inline int get_blc(struct i2s_dai *i2s) /* TX Channel Control */ static void i2s_txctrl(struct i2s_dai *i2s, int on) { - void __iomem *addr = i2s->addr; + struct samsung_i2s_priv *priv = i2s->priv; + void __iomem *addr = priv->addr; int txr_off = i2s->variant_regs->txr_off; u32 con = readl(addr + I2SCON); u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); @@ -448,7 +459,8 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on) /* RX Channel Control */ static void i2s_rxctrl(struct i2s_dai *i2s, int on) { - void __iomem *addr = i2s->addr; + struct samsung_i2s_priv *priv = i2s->priv; + void __iomem *addr = priv->addr; int txr_off = i2s->variant_regs->txr_off; u32 con = readl(addr + I2SCON); u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); @@ -485,9 +497,9 @@ static inline void i2s_fifo(struct i2s_dai *i2s, u32 flush) return; if (is_secondary(i2s)) - fic = i2s->addr + I2SFICS; + fic = i2s->priv->addr + I2SFICS; else - fic = i2s->addr + I2SFIC; + fic = i2s->priv->addr + I2SFIC; /* Flush the FIFO */ writel(readl(fic) | flush, fic); @@ -516,7 +528,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, pm_runtime_get_sync(dai->dev); spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); switch (clk_id) { @@ -613,9 +625,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs, } spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); mod = (mod & ~mask) | val; - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); done: pm_runtime_put(dai->dev); @@ -698,7 +710,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) pm_runtime_get_sync(dai->dev); spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); /* * Don't change the I2S mode if any controller is active on this * channel. @@ -714,7 +726,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); pm_runtime_put(dai->dev); @@ -801,9 +813,9 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } spin_lock_irqsave(i2s->lock, flags); - mod = readl(i2s->addr + I2SMOD); + mod = readl(priv->addr + I2SMOD); mod = (mod & ~mask) | val; - writel(mod, i2s->addr + I2SMOD); + writel(mod, priv->addr + I2SMOD); spin_unlock_irqrestore(i2s->lock, flags); snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); @@ -837,7 +849,7 @@ static int i2s_startup(struct snd_pcm_substream *substream, i2s->mode |= DAI_MANAGER; if (!any_active(i2s) && (i2s->quirks & QUIRK_NEED_RSTCLR)) - writel(CON_RSTCLR, i2s->addr + I2SCON); + writel(CON_RSTCLR, i2s->priv->addr + I2SCON); spin_unlock_irqrestore(&lock, flags); @@ -920,7 +932,7 @@ static int config_setup(struct i2s_dai *i2s) if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { psr = priv->rclk_srcrate / i2s->frmclk / rfs; - writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); + writel(((psr - 1) << 8) | PSR_PSREN, priv->addr + I2SPSR); dev_dbg(&i2s->pdev->dev, "RCLK_SRC=%luHz PSR=%u, RCLK=%dfs, BCLK=%dfs\n", priv->rclk_srcrate, psr, rfs, bfs); @@ -1008,8 +1020,9 @@ static int i2s_set_clkdiv(struct snd_soc_dai *dai, static snd_pcm_sframes_t i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); - u32 reg = readl(i2s->addr + I2SFIC); + u32 reg = readl(priv->addr + I2SFIC); snd_pcm_sframes_t delay; const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; @@ -1018,7 +1031,7 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) delay = FIC_RXCOUNT(reg); else if (is_secondary(i2s)) - delay = FICS_TXCOUNT(readl(i2s->addr + I2SFICS)); + delay = FICS_TXCOUNT(readl(priv->addr + I2SFICS)); else delay = (reg >> i2s_regs->ftx0cnt_off) & 0x7f; @@ -1042,6 +1055,7 @@ static int i2s_resume(struct snd_soc_dai *dai) static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = get_other_dai(i2s); unsigned long flags; @@ -1056,10 +1070,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) &i2s->dma_capture); if (i2s->quirks & QUIRK_NEED_RSTCLR) - writel(CON_RSTCLR, i2s->addr + I2SCON); + writel(CON_RSTCLR, priv->addr + I2SCON); if (i2s->quirks & QUIRK_SUPPORTS_IDMA) - idma_reg_addr_init(i2s->addr, + idma_reg_addr_init(priv->addr, i2s->sec_dai->idma_playback.addr); } @@ -1086,6 +1100,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) { + struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); struct i2s_dai *i2s = to_info(dai); unsigned long flags; @@ -1094,7 +1109,7 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) if (!is_secondary(i2s)) { if (i2s->quirks & QUIRK_NEED_RSTCLR) { spin_lock_irqsave(i2s->lock, flags); - writel(0, i2s->addr + I2SCON); + writel(0, priv->addr + I2SCON); spin_unlock_irqrestore(i2s->lock, flags); } } @@ -1205,11 +1220,10 @@ static int i2s_alloc_dais(struct samsung_i2s_priv *priv, static int i2s_runtime_suspend(struct device *dev) { struct samsung_i2s_priv *priv = dev_get_drvdata(dev); - struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); - priv->suspend_i2smod = readl(i2s->addr + I2SMOD); - priv->suspend_i2scon = readl(i2s->addr + I2SCON); - priv->suspend_i2spsr = readl(i2s->addr + I2SPSR); + priv->suspend_i2smod = readl(priv->addr + I2SMOD); + priv->suspend_i2scon = readl(priv->addr + I2SCON); + priv->suspend_i2spsr = readl(priv->addr + I2SPSR); if (priv->op_clk) clk_disable_unprepare(priv->op_clk); @@ -1221,7 +1235,6 @@ static int i2s_runtime_suspend(struct device *dev) static int i2s_runtime_resume(struct device *dev) { struct samsung_i2s_priv *priv = dev_get_drvdata(dev); - struct i2s_dai *i2s = samsung_i2s_get_pri_dai(dev); int ret; ret = clk_prepare_enable(priv->clk); @@ -1236,9 +1249,9 @@ static int i2s_runtime_resume(struct device *dev) } } - writel(priv->suspend_i2scon, i2s->addr + I2SCON); - writel(priv->suspend_i2smod, i2s->addr + I2SMOD); - writel(priv->suspend_i2spsr, i2s->addr + I2SPSR); + writel(priv->suspend_i2scon, priv->addr + I2SCON); + writel(priv->suspend_i2smod, priv->addr + I2SMOD); + writel(priv->suspend_i2spsr, priv->addr + I2SPSR); return 0; } @@ -1296,21 +1309,21 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { /* Activate the prescaler */ - u32 val = readl(i2s->addr + I2SPSR); - writel(val | PSR_PSREN, i2s->addr + I2SPSR); + u32 val = readl(priv->addr + I2SPSR); + writel(val | PSR_PSREN, priv->addr + I2SPSR); priv->clk_table[CLK_I2S_RCLK_SRC] = clk_register_mux(dev, i2s_clk_name[CLK_I2S_RCLK_SRC], p_names, ARRAY_SIZE(p_names), CLK_SET_RATE_NO_REPARENT | CLK_SET_RATE_PARENT, - i2s->addr + I2SMOD, reg_info->rclksrc_off, + priv->addr + I2SMOD, reg_info->rclksrc_off, 1, 0, i2s->lock); priv->clk_table[CLK_I2S_RCLK_PSR] = clk_register_divider(dev, i2s_clk_name[CLK_I2S_RCLK_PSR], i2s_clk_name[CLK_I2S_RCLK_SRC], CLK_SET_RATE_PARENT, - i2s->addr + I2SPSR, 8, 6, 0, i2s->lock); + priv->addr + I2SPSR, 8, 6, 0, i2s->lock); p_names[0] = i2s_clk_name[CLK_I2S_RCLK_PSR]; priv->clk_data.clk_num = 2; @@ -1319,7 +1332,7 @@ static int i2s_register_clock_provider(struct samsung_i2s_priv *priv) priv->clk_table[CLK_I2S_CDCLK] = clk_register_gate(dev, i2s_clk_name[CLK_I2S_CDCLK], p_names[0], CLK_SET_RATE_PARENT, - i2s->addr + I2SMOD, reg_info->cdclkcon_off, + priv->addr + I2SMOD, reg_info->cdclkcon_off, CLK_GATE_SET_TO_DISABLE, i2s->lock); priv->clk_data.clk_num += 1; @@ -1423,9 +1436,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(pri_dai->addr)) - return PTR_ERR(pri_dai->addr); + priv->addr = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(priv->addr)) + return PTR_ERR(priv->addr); regs_base = res->start; @@ -1472,7 +1485,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.addr_width = 4; - sec_dai->addr = pri_dai->addr; sec_dai->quirks = quirks; sec_dai->idma_playback.addr = idma_addr; sec_dai->pri_dai = pri_dai; -- 2.20.1

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