March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: topology: create tlv before soc_tplg_init_kcontrol
by Bard liao 15 Mar '19

15 Mar '19

Component driver may want to use tlv data. Create tlv before soc_tplg_init_kcontrol so component driver can use the tlv data in the control_load ops. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-topology.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 25fca7055464..fbee4720b834 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -894,19 +894,20 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc, &mc->hdr); + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, (struct snd_soc_tplg_ctl_hdr *) mc); if (err < 0) { dev_err(tplg->dev, "ASoC: failed to init %s\n", mc->hdr.name); + soc_tplg_free_tlv(tplg, &kc); kfree(sm); continue; } - /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->hdr); - /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, &sm->dobj.control.kcontrol); @@ -1324,18 +1325,19 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], (struct snd_soc_tplg_ctl_hdr *)mc); if (err < 0) { dev_err(tplg->dev, "ASoC: failed to init %s\n", mc->hdr.name); + soc_tplg_free_tlv(tplg, &kc[i]); kfree(sm); continue; } - - /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); } return kc; -- 2.17.1

3 2

[alsa-devel] Playing a wav file for a tone in an embedded project
by Jeff Horn 15 Mar '19

15 Mar '19

Hi, I am working on an embedded project ( i.MX6 ) and having an issue using a wav file for a tone in a Qt application. I am upgrading an existing project from Qt 5.3 with alsa-lib 1.0.8 and a 3.10.17 kernel to Qt 5.11 with alsa-lib 1.1.6 and a 4.9.123 kernel. The beep code is pretty simple, it loads a wav file and saves the wav data in a buffer. Then on each click event in Qt a beep() function is called. The beep() function calls snd_pcm_prepare() -> snd_pcm_writei() -> snd_pcm_drain(). With the previous release the entire beep() function takes ~160ms with the snd_pcm_prepare() close to 0ms. However, with the latest release the beep() function takes 550ms with 400ms in snd_pcm_prepare(). I tried to trace snd_pcm_prepare() in the kernel and found that the code path lands in dapm_power_widgets() and this is where a majority of the prepare call path time is spent. I don't know why the previous release seems to almost bypass snd_pcm_prepare() and the newer code base does not. So, is there a way to load a wav file on startup, save the wav data and then on each invocation of a beep() function play the data without having to call prepare each time? Thanks in advance for any suggestions/help Cheers, Jeff

1 0

[alsa-devel] [PATCH] ASoC: nau8810: fix arm-linux-gnueabi-gcc 7.2.0 compile error
by John Hsu 15 Mar '19

15 Mar '19

The patch fixes the compile error at the branch 'asoc for-5.2' as follows: compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 ERROR: "__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined! Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 5d300b790f78..ede2030b987a 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -21,6 +21,7 @@ #include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/math64.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -497,6 +498,7 @@ static int nau8810_calc_pll(unsigned int pll_in, unsigned int fs, struct nau8810_pll *pll_param) { u64 f2, f2_max, pll_ratio; + u32 mclk_scaler; int i, scal_sel; if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) @@ -505,7 +507,8 @@ static int nau8810_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 = 256ULL * fs * 4 * nau8810_mclk_scaler[i] / 10; + mclk_scaler = 4 * nau8810_mclk_scaler[i] / 10; + f2 = 256ULL * fs * mclk_scaler; if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && f2_max < f2) { f2_max = f2; -- 2.18.0

1 0

[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

1 0

[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

1 0

[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

4 5

[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

4 12

[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

2 12

[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

4 5

[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

2 7