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

[alsa-devel] Applied "ASoC: fsl_audmix: remove "model" attribute" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: fsl_audmix: remove "model" attribute 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 f2a36a78423ee83b1aac79956891fcad0e833885 Mon Sep 17 00:00:00 2001 From: Viorel Suman <viorel.suman(a)nxp.com> Date: Wed, 10 Apr 2019 11:06:36 +0000 Subject: [PATCH] ASoC: fsl_audmix: remove "model" attribute Use "of_device_id.data" to specify the machine driver instead of "model" DTS attribute. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_audmix.c | 43 ++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index dabde0342c95..dc802d5c4ccd 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -445,13 +445,29 @@ static const struct regmap_config fsl_audmix_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct of_device_id fsl_audmix_ids[] = { + { + .compatible = "fsl,imx8qm-audmix", + .data = "imx-audmix", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_audmix_ids); + static int fsl_audmix_probe(struct platform_device *pdev) { struct fsl_audmix *priv; struct resource *res; + const char *mdrv; + const struct of_device_id *of_id; void __iomem *regs; int ret; - const char *sprop; + + of_id = of_match_device(fsl_audmix_ids, &pdev->dev); + if (!of_id || !of_id->data) + return -EINVAL; + + mdrv = of_id->data; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) @@ -487,19 +503,12 @@ static int fsl_audmix_probe(struct platform_device *pdev) return ret; } - sprop = of_get_property(pdev->dev.of_node, "model", NULL); - if (sprop) { - priv->pdev = platform_device_register_data(&pdev->dev, sprop, 0, - NULL, 0); - if (IS_ERR(priv->pdev)) { - ret = PTR_ERR(priv->pdev); - dev_err(&pdev->dev, - "failed to register platform %s: %d\n", sprop, - ret); - } - } else { - dev_err(&pdev->dev, "[model] attribute missing.\n"); - ret = -EINVAL; + priv->pdev = platform_device_register_data(&pdev->dev, mdrv, 0, NULL, + 0); + if (IS_ERR(priv->pdev)) { + ret = PTR_ERR(priv->pdev); + dev_err(&pdev->dev, "failed to register platform %s: %d\n", + mdrv, ret); } return ret; @@ -553,12 +562,6 @@ static const struct dev_pm_ops fsl_audmix_pm = { pm_runtime_force_resume) }; -static const struct of_device_id fsl_audmix_ids[] = { - { .compatible = "fsl,imx8qm-audmix", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, fsl_audmix_ids); - static struct platform_driver fsl_audmix_driver = { .probe = fsl_audmix_probe, .remove = fsl_audmix_remove, -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: change supported formats of DL2 and UL1
by Tzung-Bi Shih 02 May '19

02 May '19

DL2 and UL1 are for BTSCO. They should only provide 16-bit, mono, 8kHz and 16kHz to userspace. Change the formats accordingly. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 681bedb524f9..4e44e5689d6f 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -69,6 +69,46 @@ mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { {"IT6505_8CH", NULL, "TDM"}, }; +static int +mt8183_mt6358_ts3a227_max98357_bt_sco_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 8000, 16000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 1, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 1; + 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); + + return 0; +} + +static const struct snd_soc_ops mt8183_mt6358_ts3a227_max98357_bt_sco_ops = { + .startup = mt8183_mt6358_ts3a227_max98357_bt_sco_startup, +}; + static struct snd_soc_dai_link mt8183_mt6358_ts3a227_max98357_dai_links[] = { /* FE */ @@ -93,6 +133,7 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_mt6358_ts3a227_max98357_bt_sco_ops, }, { .name = "Playback_3", @@ -115,6 +156,7 @@ mt8183_mt6358_ts3a227_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_mt6358_ts3a227_max98357_bt_sco_ops, }, { .name = "Capture_2", -- 2.21.0.392.gf8f6787159e-goog

2 2

[alsa-devel] [PATCH v2 0/1] ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3
by Shunli Wang 02 May '19

02 May '19

ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 Shunli Wang (1): ASoC: Mediatek: MT8183: add I2S2 control path from UL2 and UL3 sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 38 ++++++++++++++++++++++ 1 file changed, 38 insertions(+) -- 2.18.0

2 4

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Fix potential uninitialized variable
by Annaliese McDermond 02 May '19

02 May '19

Fix compiler warning about uninitialized variable reported by Stephen Rothwell <sfr(a)canb.auug.org.au>. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4-clk.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 9e4899eb1d8e..156c153c12ab 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -82,7 +82,7 @@ static int clk_aic32x4_pll_get_muldiv(struct clk_aic32x4 *pll, int ret; ret = regmap_read(pll->regmap, AIC32X4_PLLPR, &val); - if (ret) + if (ret < 0) return ret; settings->r = val & AIC32X4_PLL_R_MASK; settings->p = (val & AIC32X4_PLL_P_MASK) >> AIC32X4_PLL_P_SHIFT; -- 2.19.1

2 2

[alsa-devel] [PATCH v4 0/4] ASoC: fsl: audmix: remove "model" attribute and fix ref leaks
by Viorel Suman 02 May '19

02 May '19

The latest audmix patch-set (v5) had the "model" attribute removed as requested by Nicolin Chen, but looks like (v4) version of DAI driver reached "for-next" branch - fix this by removing "model" attribute. Asside of this fix object reference leaks in machine probe reported by Julia Lawall. Viorel Suman (4): ASoC: fsl_audmix: remove "model" attribute ASoC: fsl_audmix: remove "model" attribute from DT document ASoC: imx-audmix: fix object reference leaks in probe ASoC: fsl_audmix: cache pdev->dev pointer Changes since V1: a) Removed "model" attribute from dt-bindings documentation b) Adressed Daniel's comments Changes since V2: a) Cache pdev->dev pointer in fsl_audmix probe as suggested by Nicolin Changes since V3: a) Use subject lines matching the style for the subsystem. .../devicetree/bindings/sound/fsl,audmix.txt | 4 -- sound/soc/fsl/fsl_audmix.c | 60 +++++++++++----------- sound/soc/fsl/imx-audmix.c | 4 ++ 3 files changed, 35 insertions(+), 33 deletions(-) -- 2.7.4

4 10

[alsa-devel] [PATCH] ASoC: dapm: fix kcontrols for effect widgets
by Pierre-Louis Bossart 02 May '19

02 May '19

From: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> This patch adds the handling of snd_soc_dapm_effect that was missing. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 65ee0bb5dd0b..81a7a12196ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -883,6 +883,7 @@ static int dapm_create_or_share_kcontrol(struct snd_soc_dapm_widget *w, case snd_soc_dapm_switch: case snd_soc_dapm_mixer: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: wname_in_long_name = true; kcname_in_long_name = true; @@ -2370,6 +2371,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_dac: case snd_soc_dapm_adc: case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: @@ -3197,6 +3199,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_new_mux(w); break; case snd_soc_dapm_pga: + case snd_soc_dapm_effect: case snd_soc_dapm_out_drv: dapm_new_pga(w); break; -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: da7213: fix DAI_CLK_EN register bit overwrite
by Logesh Kolandavel 02 May '19

02 May '19

From: Logesh <logesh.kolandavel(a)timesys.com> If the da7213 codec is configured as Master with the DAPM power down delay time set, 'snd_soc_component_write' function overwrites the DAI_CLK_EN bit of DAI_CLK_MODE register which leads to audio play only once until it re-initialize after codec power up. Signed-off-by: Logesh <logesh.kolandavel(a)timesys.com> --- Changes in v2: -Include the mask DA7213_DAI_BCLKS_PER_WCLK_MASK sound/soc/codecs/da7213.c | 5 ++++- sound/soc/codecs/da7213.h | 2 ++ 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 92d006a5283e..425c11d63e49 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1305,7 +1305,10 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* By default only 64 BCLK per WCLK is supported */ dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64; - snd_soc_component_write(component, DA7213_DAI_CLK_MODE, dai_clk_mode); + snd_soc_component_update_bits(component, DA7213_DAI_CLK_MODE, + DA7213_DAI_BCLKS_PER_WCLK_MASK | + DA7213_DAI_CLK_POL_MASK | DA7213_DAI_WCLK_POL_MASK, + dai_clk_mode); snd_soc_component_update_bits(component, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, dai_ctrl); snd_soc_component_write(component, DA7213_DAI_OFFSET, dai_offset); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 5a78dba1dcb5..9d31efc3cfe5 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -181,7 +181,9 @@ #define DA7213_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) +#define DA7213_DAI_CLK_POL_MASK (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7213_DAI_WCLK_POL_MASK (0x1 << 3) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ -- 2.21.0

3 2

[alsa-devel] [PATCH -next] ASoC: sprd: Fix to use list_for_each_entry_safe() when delete items
by Wei Yongjun 02 May '19

02 May '19

Since we will remove items off the list using list_del() we need to use a safe version of the list_for_each_entry() macro aptly named list_for_each_entry_safe(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..df250f7f2b6f 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -978,12 +978,12 @@ static int sprd_mcdt_probe(struct platform_device *pdev) static int sprd_mcdt_remove(struct platform_device *pdev) { - struct sprd_mcdt_chan *temp; + struct sprd_mcdt_chan *chan, *temp; mutex_lock(&sprd_mcdt_list_mutex); - list_for_each_entry(temp, &sprd_mcdt_chan_list, list) - list_del(&temp->list); + list_for_each_entry_safe(chan, temp, &sprd_mcdt_chan_list, list) + list_del(&chan->list); mutex_unlock(&sprd_mcdt_list_mutex);

3 2

[alsa-devel] [PATCH -next] ASoC: sprd: Fix return value check in sprd_mcdt_probe()
by Wei Yongjun 02 May '19

02 May '19

In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Fixes: d7bff893e04f ("ASoC: sprd: Add Spreadtrum multi-channel data transfer support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sprd/sprd-mcdt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 28f5e649733d..e9318d7a4810 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -951,8 +951,8 @@ static int sprd_mcdt_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); mcdt->base = devm_ioremap_resource(&pdev->dev, res); - if (!mcdt->base) - return -ENOMEM; + if (IS_ERR(mcdt->base)) + return PTR_ERR(mcdt->base); mcdt->dev = &pdev->dev; spin_lock_init(&mcdt->lock);

3 2

[alsa-devel] [PATCH][next] ASoC: SOF: Intel: fix spelling mistake "incompatble" -> "incompatible"
by Colin King 02 May '19

02 May '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a hda_dsp_rom_msg message, fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b8fc19790f3b..84baf275b467 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -54,7 +54,7 @@ static const struct hda_dsp_msg_code hda_dsp_rom_msg[] = { {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"}, {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"}, {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"}, - {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatble"}, + {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"}, {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"}, {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"}, {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"}, -- 2.20.1

4 3