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

[alsa-devel] Applied "ASoC: rename functions that pollute the simple_xxx namespace" to the asoc tree
by Mark Brown 21 May '19

21 May '19

The patch ASoC: rename functions that pollute the simple_xxx namespace has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 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 b0a821daf0d04e5a8ae99829e24f2fe538f25763 Mon Sep 17 00:00:00 2001 From: Amir Goldstein <amir73il(a)gmail.com> Date: Thu, 16 May 2019 13:26:28 +0300 Subject: [PATCH] ASoC: rename functions that pollute the simple_xxx namespace include/linux/fs.h defines a bunch of simple fs helpers, (e.g. simple_rename) and we intend to add an fs helper named simple_remove. Rename the ASoC driver static functions, so they will not collide with the upcoming fs helper function name. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Amir Goldstein <amir73il(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9b568f578bcd..d16e894fce2b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -607,7 +607,7 @@ static int simple_soc_probe(struct snd_soc_card *card) return 0; } -static int simple_probe(struct platform_device *pdev) +static int asoc_simple_probe(struct platform_device *pdev) { struct asoc_simple_priv *priv; struct device *dev = &pdev->dev; @@ -705,7 +705,7 @@ static int simple_probe(struct platform_device *pdev) return ret; } -static int simple_remove(struct platform_device *pdev) +static int asoc_simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); @@ -726,8 +726,8 @@ static struct platform_driver asoc_simple_card = { .pm = &snd_soc_pm_ops, .of_match_table = simple_of_match, }, - .probe = simple_probe, - .remove = simple_remove, + .probe = asoc_simple_probe, + .remove = asoc_simple_remove, }; module_platform_driver(asoc_simple_card); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt5677-spi: Handle over reading when flipping bytes
by Curtis Malainey 21 May '19

21 May '19

There is a case when a we want to read a large number of bytes that require a burst but is not a multiple of the word size (8). When this happens rt5677_spi_reverse will run off the end of the buffer. The solution is to tell spi_reverse the actual size of the destination and stop if we reach it even if we have data left that we read. Cc: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84b6bd8b50e1..a4dfa0345c6e 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -101,7 +101,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) u32 word_size = min_t(u32, dstlen, 8); for (w = 0; w < dstlen; w += word_size) { - for (i = 0; i < word_size; i++) { + for (i = 0; i < word_size && i + w < dstlen; i++) { si = w + word_size - i - 1; dst[w + i] = si < srclen ? src[si] : 0; } @@ -152,8 +152,9 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) status |= spi_sync(g_spi, &m); mutex_unlock(&spi_mutex); + /* Copy data back to caller buffer */ - rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len); + rt5677_spi_reverse(cb + offset, len - offset, body, t[1].len); } return status; } -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [PATCH][RFC] ASoC: soc-dpm: fixup DAI active unbalance
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_link_event() is updating snd_soc_dai :: active, but it is unbalance. It counts up if it has startup callback. case SND_SOC_DAPM_PRE_PMU: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... if (source->driver->ops->startup) { ... => source->active++; } ... } ... But, always counts down case SND_SOC_DAPM_PRE_PMD: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... => source->active--; ... } This patch always counts up when SND_SOC_DAPM_PRE_PMD. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Liam I think this is bug, but I can't confirm it, because my driver need to have .startup. Thus, I added [RFC] on this patch. I'm happy if someone can confirm it. sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c864502..147ad9d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,8 +3828,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - source->active++; } + source->active++; ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; @@ -3850,8 +3850,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - sink->active++; } + sink->active++; ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: Mediatek: MT8183: enable IIR filter
by Shunli Wang 21 May '19

21 May '19

IIR fileter can remove DC offset. It must be enabled when dmic or amic connected to pmic is used. Signed-off-by: Shunli Wang <shunli.wang(a)mediatek.com> --- v2: correct the commit message --- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 017d7d1d9148..2b758a18c2ea 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -176,9 +176,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; @@ -426,6 +423,17 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); + /* enable iir */ + ul_src_con0 |= (1 << UL_IIR_ON_TMP_CTL_SFT) & + UL_IIR_ON_TMP_CTL_MASK_SFT; + + /* 35Hz @ 48k */ + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_02_01, 0x00000000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_04_03, 0x00003FB8); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_06_05, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_08_07, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_10_09, 0x0000C048); + regmap_write(afe->regmap, AFE_ADDA_UL_SRC_CON0, ul_src_con0); /* mtkaif_rxif_data_mode = 0, amic */ -- 2.18.0

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[alsa-devel] [PATCH 0/3] ASoC: Intel: fix rt5682-based machine drivers
by Pierre-Louis Bossart 21 May '19

21 May '19

We have two machine drivers based on rt5682 and they both need a correction for the mapping of the buttons, the initial mapping taken from the DA7219 is incorrect. Also fix ACPI tables, quirks as needed. Sathya Prakash M R (2): ASoC: Intel: sof-rt5682: Few minor fixes for AMP SSP and codec button ASoC: Intel: soc-acpi: Fix machine selection order Yong Zhi (1): ASoC: Intel: glk_rt5682_max98357a: Remap button control-function sound/soc/intel/boards/glk_rt5682_max98357a.c | 7 ++++--- sound/soc/intel/boards/sof_rt5682.c | 12 +++++++----- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 10 +++++----- 3 files changed, 16 insertions(+), 13 deletions(-) base-commit: d0723c9d845f9a6ff1069e5786ccda8e81566763 -- 2.20.1

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[alsa-devel] [PATCH][RFC] ASoC: soc-core: WARN() is not related to component->driver->probe
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_probe_component() has WARN() under if (component->driver->probe), but, this WARN() check is not related to .probe callback. So, it should be called at (B) instead of (A). This patch moves it out of if(). if (component->driver->probe) { ret = component->driver->probe(component); ... (A) WARN(...) } (B) WARN(...) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Pierre-Louis, Vinod, Liam I think this patch is correct, but I'm not sure. I'm happy someone can confirm it. sound/soc/soc-core.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cab30ae..7157d67 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1420,12 +1420,11 @@ static int soc_probe_component(struct snd_soc_card *card, "ASoC: failed to probe component %d\n", ret); goto err_probe; } - - WARN(dapm->idle_bias_off && - dapm->bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - component->name); } + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); /* machine specific init */ if (component->init) { -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: move pcm_new from snd_soc_component_driver to snd_soc_dai_driver
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_driver :: pcm_new has snd_soc_dai as parameter, but snd_soc_component_driver :: pcm_new doesn't have it. rsnd driver needs snd_soc_dai at pcm_new. This patch moves .pcm_new from snd_soc_component_driver to snd_soc_dai_driver, and don't use rtd->cpu_dai anymore. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 120 +++++++++++++++++++++++------------------------ 1 file changed, 60 insertions(+), 60 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 37cb615..56e8dae 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1176,6 +1176,65 @@ static struct device_node *rsnd_dai_of_node(struct rsnd_priv *priv, return ret; } + +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static int rsnd_preallocate_pages(struct snd_soc_pcm_runtime *rtd, + struct rsnd_dai_stream *io, + int stream) +{ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_pcm_substream *substream; + + /* + * use Audio-DMAC dev if we can use IPMMU + * see + * rsnd_dmaen_attach() + */ + if (io->dmac_dev) + dev = io->dmac_dev; + + for (substream = rtd->pcm->streams[stream].substream; + substream; + substream = substream->next) { + snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + dev, + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); + } + + return 0; +} + +static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + int ret; + + ret = rsnd_dai_call(pcm_new, &rdai->playback, rtd); + if (ret) + return ret; + + ret = rsnd_dai_call(pcm_new, &rdai->capture, rtd); + if (ret) + return ret; + + ret = rsnd_preallocate_pages(rtd, &rdai->playback, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + + ret = rsnd_preallocate_pages(rtd, &rdai->capture, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + + return 0; +} + static void __rsnd_dai_probe(struct rsnd_priv *priv, struct device_node *dai_np, int dai_i) @@ -1198,6 +1257,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, rdai->priv = priv; drv->name = rdai->name; drv->ops = &rsnd_soc_dai_ops; + drv->pcm_new = rsnd_pcm_new; snprintf(io_playback->name, RSND_DAI_NAME_SIZE, "DAI%d Playback", dai_i); @@ -1572,68 +1632,8 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, /* * snd_soc_component */ - -#define PREALLOC_BUFFER (32 * 1024) -#define PREALLOC_BUFFER_MAX (32 * 1024) - -static int rsnd_preallocate_pages(struct snd_soc_pcm_runtime *rtd, - struct rsnd_dai_stream *io, - int stream) -{ - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct device *dev = rsnd_priv_to_dev(priv); - struct snd_pcm_substream *substream; - - /* - * use Audio-DMAC dev if we can use IPMMU - * see - * rsnd_dmaen_attach() - */ - if (io->dmac_dev) - dev = io->dmac_dev; - - for (substream = rtd->pcm->streams[stream].substream; - substream; - substream = substream->next) { - snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - dev, - PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); - } - - return 0; -} - -static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *dai = rtd->cpu_dai; - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - int ret; - - ret = rsnd_dai_call(pcm_new, &rdai->playback, rtd); - if (ret) - return ret; - - ret = rsnd_dai_call(pcm_new, &rdai->capture, rtd); - if (ret) - return ret; - - ret = rsnd_preallocate_pages(rtd, &rdai->playback, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - return ret; - - ret = rsnd_preallocate_pages(rtd, &rdai->capture, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - return ret; - - return 0; -} - static const struct snd_soc_component_driver rsnd_soc_component = { .ops = &rsnd_pcm_ops, - .pcm_new = rsnd_pcm_new, .name = "rsnd", }; -- 2.7.4

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[alsa-devel] [PATCH v2 0/5] ASoC: Intel: fix rt5682-based machine drivers
by Pierre-Louis Bossart 21 May '19

21 May '19

We have two machine drivers based on rt5682 and they both need a correction for the mapping of the buttons, the initial mapping taken from the DA7219 is incorrect. Also fix ACPI tables, quirks as needed, and add newlines for clarity Changes since v1: Split first patch in 3 (feedback from Mark). No code changes. Sathya Prakash M R (4): ASoC: Intel: sof-rt5682: fix for codec button mapping ASoC: Intel: sof-rt5682: fix AMP quirk support ASoC: Intel: sof-rt5682: add newline for clarity ASoC: Intel: soc-acpi: Fix machine selection order Yong Zhi (1): ASoC: Intel: glk_rt5682_max98357a: Remap button control-function sound/soc/intel/boards/glk_rt5682_max98357a.c | 7 ++++--- sound/soc/intel/boards/sof_rt5682.c | 12 +++++++----- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 10 +++++----- 3 files changed, 16 insertions(+), 13 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH V5] ASoC: cs42xx8: add reset-gpios in binding document
by S.j. Wang 21 May '19

21 May '19

Add reset-gpios property, which is optional. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in V5 - use GPIO_ACTIVE_LOW Changes in V4 - use gpios instead of gpio Changes in RESEND v3 - send updated binding document only Changes in v3 - update binding document. Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio Documentation/devicetree/bindings/sound/cs42xx8.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt index 8619a156d038..bbfe39347c20 100644 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -14,6 +14,11 @@ Required properties: - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - reset-gpios : a GPIO spec to define which pin is connected to the chip's + !RESET pin + Example: cs42888: codec@48 { @@ -25,4 +30,5 @@ cs42888: codec@48 { VD-supply = <&reg_audio>; VLS-supply = <&reg_audio>; VLC-supply = <&reg_audio>; + reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>; }; -- 2.21.0

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[alsa-devel] [PATCH 0/2] ASoC: Mediatek: MT8183: fix compile error
by Jiaxin Yu 21 May '19

21 May '19

This patch fixed the compile error which patches for support S32_LE and S24_LE. The patch is based on broonie "for-5.3" branch. Jiaxin Yu (2): ASoC: Mediatek: MT8183: fix compile error ASoC: Mediatek: MT8183: fix compile error sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +-- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 18 +++++++++--------- 2 files changed, 10 insertions(+), 11 deletions(-) -- 1.8.1.1.dirty

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