April 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.6 119/149] ASoC: stm32: spdifrx: fix regmap status check
by Sasha Levin 12 Apr '20

12 Apr '20

From: Olivier Moysan <olivier.moysan(a)st.com> [ Upstream commit a168dae5ea14283e8992d5282237bb0d6a3e1c06 ] Release resources when exiting on error. Fixes: 1a5c0b28fc56 ("ASoC: stm32: spdifrx: manage identification registers") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Link: https://lore.kernel.org/r/20200318144125.9163-2-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_spdifrx.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 3769d9ce5dbef..e6e75897cce83 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1009,6 +1009,8 @@ static int stm32_spdifrx_probe(struct platform_device *pdev) if (idr == SPDIFRX_IPIDR_NUMBER) { ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver); + if (ret) + goto error; dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n", FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver), -- 2.20.1

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[PATCH AUTOSEL 5.6 086/149] ASoC: soc-pcm: fix regression in soc_new_pcm()
by Sasha Levin 12 Apr '20

12 Apr '20

From: Stephan Gerhold <stephan(a)gerhold.net> [ Upstream commit a4877a6fb2bd2e356a5eaacd86d6b6d69ff84e69 ] Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") swapped the SNDRV_PCM_STREAM_* parameter in the snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only for codec2codec links. For normal links it breaks registration of playback/capture-only PCM devices. E.g. on qcom/apq8016_sbc there is usually one playback-only and one capture-only PCM device, but they disappeared after the commit. The codec2codec case was added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") as an extra check (e.g. `playback = playback && cpu_playback->channels_min`). We should be able to simplify the code by checking directly for the correct stream type in the loop. This also fixes the regression because we check for PLAYBACK for both codec and cpu dai again when codec2codec is not used. Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> Tested-by: Jerome Brunet <jbrunet(a)baylibre.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Cc: Jerome Brunet <jbrunet(a)baylibre.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Link: https://lore.kernel.org/r/20200218103824.26708-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca01..a08ebbf5a37f3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2888,22 +2888,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { /* Adapt stream for codec2codec links */ - struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ? - &cpu_dai->driver->playback : &cpu_dai->driver->capture; - struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ? - &cpu_dai->driver->capture : &cpu_dai->driver->playback; + int cpu_capture = rtd->dai_link->params ? + SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int cpu_playback = rtd->dai_link->params ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, cpu_capture)) capture = 1; } - - capture = capture && cpu_capture->channels_min; - playback = playback && cpu_playback->channels_min; } if (rtd->dai_link->playback_only) { -- 2.20.1

1 0

[PATCH AUTOSEL 5.6 081/149] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed
by Sasha Levin 12 Apr '20

12 Apr '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d ] 1. Switch TDM GPIO pins according to playback on or off. 2. Pull TDM GPIO pins down when probed to avoid current leakage. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++- 1 file changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c65493721e90d..a3974b0a90157 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,6 +18,22 @@ static struct snd_soc_jack headset_jack; +enum PINCTRL_PIN_STATE { + PIN_STATE_DEFAULT = 0, + PIN_TDM_OUT_ON, + PIN_TDM_OUT_OFF, + PIN_STATE_MAX +}; + +static const char * const mt8183_pin_str[PIN_STATE_MAX] = { + "default", "aud_tdm_out_on", "aud_tdm_out_off", +}; + +struct mt8183_da7219_max98357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -204,6 +220,47 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) + return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + + return ret; +} + +static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); +} + +static struct snd_soc_ops mt8183_da7219_tdm_ops = { + .startup = mt8183_da7219_tdm_startup, + .shutdown = mt8183_da7219_tdm_shutdown, +}; + static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -353,6 +410,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -409,7 +468,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; int ret, i; card->dev = &pdev->dev; @@ -443,12 +502,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + priv->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(priv->pinctrl)) { + dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(default_pins); + return PTR_ERR(priv->pinctrl); + } + + for (i = 0; i < PIN_STATE_MAX; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8183_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", + __func__, mt8183_pin_str[i], ret); + } + } + + if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); + } + + if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_DEFAULT]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); } return ret; -- 2.20.1

1 0

[PATCH] ALSA: hda: add autodetection for SoundWire
by Pierre-Louis Bossart 11 Apr '20

11 Apr '20

When an ACPI companion device is present and the SoundWire link mask information is available, use SoundWire instead of legacy HDA or Skylake drivers. The SOF driver is selected when SoundWire or DMIC are detected. There is no precedence at this level. In the SOF driver proper, SoundWire will be handled first since it is mutually exclusive with HDaudio. Known devices with an existing DMI quirk bypass this detection to avoid any dependency on ACPI/DSDT tables. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 97 ++++++++++++++++++++++++++++++------ 1 file changed, 81 insertions(+), 16 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index be1df80ed013..1c1fdab2a55f 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -1,10 +1,13 @@ // SPDX-License-Identifier: GPL-2.0 // Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> +#include <linux/acpi.h> #include <linux/bits.h> #include <linux/dmi.h> #include <linux/module.h> #include <linux/pci.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/core.h> #include <sound/intel-dsp-config.h> #include <sound/intel-nhlt.h> @@ -14,9 +17,13 @@ static int dsp_driver; module_param(dsp_driver, int, 0444); MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); -#define FLAG_SST BIT(0) -#define FLAG_SOF BIT(1) -#define FLAG_SOF_ONLY_IF_DMIC BIT(16) +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) +#define FLAG_SOF_ONLY_IF_SOUNDWIRE BIT(17) + +#define FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE (FLAG_SOF_ONLY_IF_DMIC | \ + FLAG_SOF_ONLY_IF_SOUNDWIRE) struct config_entry { u32 flags; @@ -166,7 +173,7 @@ static const struct config_entry config_table[] = { } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x9dc8, }, #endif @@ -187,7 +194,7 @@ static const struct config_entry config_table[] = { } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa348, }, #endif @@ -204,18 +211,50 @@ static const struct config_entry config_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), } }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "09C6") + }, + }, + { + /* early version of SKU 09C6 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0983") + }, + }, {} } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x02c8, }, #endif /* Cometlake-H */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF, + .device = 0x06c8, + .dmi_table = (const struct dmi_system_id []) { + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"), + }, + }, + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"), + }, + }, + {} + } + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x06c8, }, #endif @@ -236,7 +275,7 @@ static const struct config_entry config_table[] = { } }, { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x34c8, }, #endif @@ -256,9 +295,8 @@ static const struct config_entry config_table[] = { {} } }, - { - .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0xa0c8, }, #endif @@ -303,6 +341,28 @@ static int snd_intel_dsp_check_dmic(struct pci_dev *pci) return ret; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int snd_intel_dsp_check_soundwire(struct pci_dev *pci) +{ + struct sdw_intel_acpi_info info; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(&pci->dev); + + ret = sdw_intel_acpi_scan(handle, &info); + if (ret < 0) + return ret; + + return info.link_mask; +} +#else +static int snd_intel_dsp_check_soundwire(struct pci_dev *pci) +{ + return 0; +} +#endif + int snd_intel_dsp_driver_probe(struct pci_dev *pci) { const struct config_entry *cfg; @@ -336,14 +396,18 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) return SND_INTEL_DSP_DRIVER_ANY; if (cfg->flags & FLAG_SOF) { - if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { - if (snd_intel_dsp_check_dmic(pci)) { - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); - return SND_INTEL_DSP_DRIVER_SOF; - } - } else { + if (cfg->flags & FLAG_SOF_ONLY_IF_SOUNDWIRE && + snd_intel_dsp_check_soundwire(pci) > 0) { + dev_info(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC && + snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); return SND_INTEL_DSP_DRIVER_SOF; } + if (!(cfg->flags & FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE)) + return SND_INTEL_DSP_DRIVER_SOF; } if (cfg->flags & FLAG_SST) @@ -355,3 +419,4 @@ EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("Intel DSP config driver"); +MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek - Enable the headset mic on Asus FX505DT
by Adam Barber 11 Apr '20

11 Apr '20

On Asus FX505DT with Realtek ALC233, the headset mic is connected to pin 0x19, with default 0x411111f0. Enable headset mic by reconfiguring the pin to an external mic associated with the headphone on 0x21. Mic jack detection was also found to be working. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207131 Signed-off-by: Adam Barber <barberadam995(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f66a48154a57..5016fa50956a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7252,6 +7252,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), -- 2.26.0

2 1

Re: [PATCH] ALSA: hda/realtek - Enable the headset mic on Asus FX505DT
by Takashi Iwai 10 Apr '20

10 Apr '20

On Fri, 10 Apr 2020 02:55:33 +0200, Adam Barber wrote: > > On Asus FX505DT with Realtek ALC233, the headset mic is connected > to pin 0x19, with default 0x411111f0. > > Enable headset mic by reconfiguring the pin to an external mic > associated with the headphone on 0x21. Mic jack detection was also > found to be working. Could you give your sign-off? It's mandatory for merging to upstream. Also, if there is some information linke (e.g. Bugzilla), put it in "BugLink: $URL" line, too. thanks, Takashi > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index f66a48154a57..5016fa50956a 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -7252,6 +7252,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), > SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), > SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), > + SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), > SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), > SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), > SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), > -- > 2.26.0 >

1 0

[PATCH] ASoC: soc-core: Remove name length truncation
by Tom Eccles 09 Apr '20

09 Apr '20

snd_soc_dai.name and snd_soc_component.name are a simplified version of the parent device's name, truncated to NAME_SIZE characters (see fmt_single_name). NAME_SIZE was 32 characters. Some device names may share a common prefix longer than 32 characters, which leads to non-unique snd_soc_dai.name and snd_soc_component.name. When these are added to debugfs, the .name is used for the file. If the names are not unique, adding to debugfs fails. Instead, dynamically allocate name buffers to be as long as needed. Signed-off-by: Tom Eccles <tom.eccles(a)codethink.co.uk> --- sound/soc/soc-core.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 843b8b1c89d4..7803222ede54 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -43,8 +43,6 @@ #define CREATE_TRACE_POINTS #include <trace/events/asoc.h> -#define NAME_SIZE 32 - static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); static LIST_HEAD(unbind_card_list); @@ -2291,13 +2289,15 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); */ static char *fmt_single_name(struct device *dev, int *id) { - char *found, name[NAME_SIZE]; + char *found, *name; int id1, id2; if (dev_name(dev) == NULL) return NULL; - strlcpy(name, dev_name(dev), NAME_SIZE); + name = devm_kstrdup(dev, dev_name(dev), GFP_KERNEL); + if (!name) + return NULL; /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); @@ -2313,20 +2313,23 @@ static char *fmt_single_name(struct device *dev, int *id) } else { /* I2C component devices are named "bus-addr" */ if (sscanf(name, "%x-%x", &id1, &id2) == 2) { - char tmp[NAME_SIZE]; + char *tmp; /* create unique ID number from I2C addr and bus */ *id = ((id1 & 0xffff) << 16) + id2; /* sanitize component name for DAI link creation */ - snprintf(tmp, NAME_SIZE, "%s.%s", dev->driver->name, - name); - strlcpy(name, tmp, NAME_SIZE); + tmp = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", + dev->driver->name, name); + devm_kfree(dev, name); + if (!tmp) + return NULL; + name = tmp; } else *id = 0; } - return devm_kstrdup(dev, name, GFP_KERNEL); + return name; } /* base-commit: fff876253c1e1b329257ba33045195c93e25adcd -- 2.20.1

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Re: [PATCH] ALSA: ctxfi: Remove unnecessary cast in kfree
by Takashi Iwai 09 Apr '20

09 Apr '20

On Thu, 09 Apr 2020 13:20:52 +0200, Xu Wang wrote: > > Remove unnecassary casts in the argument to kfree. > > Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> Applied, thanks. Takashi

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[PATCH v2 0/6] ASoC: topology: Propagate error appropriately
by Amadeusz Sławiński 09 Apr '20

09 Apr '20

v1: Check if kstrdup succeeded. v2: Remove unneeded freeing, which is performed in another place by dobj handlers. Additionally for functions which have return status which was ignored, perform success checks and handle failures in appropriate way. Amadeusz Sławiński (6): ASoC: topology: Add missing memory checks ASoC: topology: Check return value of soc_tplg_create_tlv ASoC: topology: Check return value of soc_tplg_*_create ASoC: topology: Check soc_tplg_add_route return value ASoC: topology: Check return value of pcm_new_ver ASoC: topology: Check return value of soc_tplg_dai_config sound/soc/soc-topology.c | 113 ++++++++++++++++++++++++++++++--------- 1 file changed, 88 insertions(+), 25 deletions(-) -- 2.17.1

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alsa-gobject: Add shared library to operate ALSA rawmidi character device
by GitHub pull_request - opened 09 Apr '20

09 Apr '20

alsa-project/alsa-gobject pull request #16 was opened from takaswie: This patchset is to add a new library, libalsarawmidi, to operate ALSA rawmidi character device. This library is compatible with GObject Introspection. ALSA rawmidi interface is designed to transfer MIDI messages. Userspace application operate ALSA rawmidi character device for incoming/outgoing transmission. The transmission line is abstracted as substream. One character device is associated to several substreams. The application can indicate subdevice to select a substream or a pair of substreams from them. The library has ALSARawmidi.StreamPair to operate the character device. Userspace application call ALSARawmidi.StreamPair.open() to open the chracter device, select subdevice and attach the substreams to the device. After the call, data transmission already starts. The application can call ALSARawmidi.StreamPair.write_to_substream() to transfer MIDI message bytes. The application can also call ALSARawmidi.StreamPair.read_from_substream() to receive MIDI message bytes, but this call can be blocked till any messages are actually received and available as a default. For polling, GSource which ALSARawmidi.StreamPair.create_source() returns is useful. The application can use it for GMainContext/GMainLoop. When any message is available, ALSARawmidi.StreamPair emits 'handle-messages' GObject signal. The application register its handler to the signal in advance, then process the messages. ALSARawmidi.StreamPair has the other methods to drain/drop running substream: - ALSARawmidi.StreamPair.drain_substream() - ALSARawmidi.StreamPair.drop_substream() ALSARawmidi.StreamPair has more methods to return GObject-derived objects for information/status/parameters of substream: - ALSARawmidi.StreamPair.get_substream_info() - ALSARawmidi.StreamPair.get_substream_status() - ALSARawmidi.StreamPair.set_substream_params() The library includes global method to query information about rawmidi: - ALSARawmidi.get_rawmidi_sysname() - ALSARawmidi.get_rawmidi_devnode() - ALSARawmidi.get_device_id_list() - ALSARawmidi.get_subdevice_id_list() - ALSARawmidi.get_substream_info() Request URL : https://github.com/alsa-project/alsa-gobject/pull/16 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/16.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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