August 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal 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 dba508b5ab1d138bd7543a3f503492f1a173aa32 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index c0bc429249fa..3959e6ad113d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -67,6 +67,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -320,7 +323,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -331,7 +334,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static bool adau17x1_has_safeload(struct adau *adau) { @@ -854,7 +856,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -898,7 +900,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal 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 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] [PATCH 0/1] ASoC: pcm3060: Add codec driver
by Kirill Marinushkin 29 Aug '18

29 Aug '18

Hello Mark, I am developing sound support for a Linux-based device, with playback and capture through the TI PCM3060 codec. With the following patch, I propose to add the PCM3060 codec driver into the kernel. Best Regards, Kirill Kirill Marinushkin (1): ASoC: pcm3060: Add codec driver .../devicetree/bindings/sound/pcm3060.txt | 17 ++ MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 17 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/pcm3060-i2c.c | 61 +++++ sound/soc/codecs/pcm3060-spi.c | 60 +++++ sound/soc/codecs/pcm3060.c | 290 +++++++++++++++++++++ sound/soc/codecs/pcm3060.h | 88 +++++++ 8 files changed, 546 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm3060.txt create mode 100644 sound/soc/codecs/pcm3060-i2c.c create mode 100644 sound/soc/codecs/pcm3060-spi.c create mode 100644 sound/soc/codecs/pcm3060.c create mode 100644 sound/soc/codecs/pcm3060.h -- 2.13.6

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[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal 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 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: adau17x1: Unused exported functions changed to internal" to the asoc tree
by Mark Brown 29 Aug '18

29 Aug '18

The patch ASoC: adau17x1: Unused exported functions changed to internal 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 164c0f6454186d1b0e22c27dfe3d3d59fc80f979 Mon Sep 17 00:00:00 2001 From: Robert Rosengren <robert.rosengren(a)axis.com> Date: Mon, 13 Aug 2018 09:33:58 +0200 Subject: [PATCH] ASoC: adau17x1: Unused exported functions changed to internal adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so making them static instead of exported. Change-Id: I4882513ab457966b0371ac2c1024550ca6edbe0c Signed-off-by: Robert Rosengren <robertr(a)axis.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau17x1.c | 9 +++++---- sound/soc/codecs/adau17x1.h | 4 ---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..2f2afb4c0c88 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -60,6 +60,9 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { SOC_ENUM("Mic Bias Mode", adau17x1_mic_bias_mode_enum), }; +static int adau17x1_setup_firmware(struct snd_soc_component *component, + unsigned int rate); + static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -313,7 +316,7 @@ static const struct snd_soc_dapm_route adau17x1_no_dsp_dapm_routes[] = { { "Capture", NULL, "Right Decimator" }, }; -bool adau17x1_has_dsp(struct adau *adau) +static bool adau17x1_has_dsp(struct adau *adau) { switch (adau->type) { case ADAU1761: @@ -324,7 +327,6 @@ bool adau17x1_has_dsp(struct adau *adau) return false; } } -EXPORT_SYMBOL_GPL(adau17x1_has_dsp); static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) @@ -836,7 +838,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_setup_firmware(struct snd_soc_component *component, +static int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate) { int ret; @@ -880,7 +882,6 @@ int adau17x1_setup_firmware(struct snd_soc_component *component, return ret; } -EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_component *component) { diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..98a3b6f5bc96 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -68,10 +68,6 @@ int adau17x1_resume(struct snd_soc_component *component); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_setup_firmware(struct snd_soc_component *component, - unsigned int rate); -bool adau17x1_has_dsp(struct adau *adau); - #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 #define ADAU17X1_REC_POWER_MGMT 0x4009 -- 2.18.0

1 0

[alsa-devel] [PATCH 1/3] ASoC: cs4265: SOC_SINGLE register value error fix
by Matt Flax 29 Aug '18

29 Aug '18

The cs4265 driver declares the "MMTLR Data Switch" register setting with a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2). This incorrect value causes alsamixer to fault with the output : cannot load mixer controls: Input/output error This patch corrects the register value. alsamixer now runs. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 275677de..15b4ae04 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -157,8 +157,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), - SOC_SINGLE("MMTLR Data Switch", 0, - 1, 1, 0), + SOC_SINGLE("MMTLR Data Switch", CS4265_SPDIF_CTL2, 0, 1, 0), SOC_ENUM("Mono Channel Select", spdif_mono_select_enum), SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24), }; -- 2.17.1

3 13

[alsa-devel] Applied "ASoC: max98373: Added speaker FS gain cotnrol register to volatile." to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: max98373: Added speaker FS gain cotnrol register to volatile. 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 0d22825255f25adb6a609f130b42c752d3fd0f5d Mon Sep 17 00:00:00 2001 From: Ryan Lee <ryans.lee(a)maximintegrated.com> Date: Wed, 15 Aug 2018 18:53:38 -0700 Subject: [PATCH] ASoC: max98373: Added speaker FS gain cotnrol register to volatile. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 92b7125ea169..2764fae69333 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -520,6 +520,7 @@ static bool max98373_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case MAX98373_R2000_SW_RESET ... MAX98373_R2009_INT_FLAG3: + case MAX98373_R203E_AMP_PATH_GAIN: case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: case MAX98373_R20B6_BDE_CUR_STATE_READBACK: -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: rt5514: Fix the issue of the delay volume applied again" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: rt5514: Fix the issue of the delay volume applied again 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 6f0a256253f48095ba2e5bcdfbed41f21643c105 Mon Sep 17 00:00:00 2001 From: Oder Chiou <oder_chiou(a)realtek.com> Date: Wed, 15 Aug 2018 14:47:49 +0800 Subject: [PATCH] ASoC: rt5514: Fix the issue of the delay volume applied again After our evaluation, we need to modify the default values to make sure the volume applied immediately. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index dca82dd6e3bf..32fe76c3134a 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000342}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000342}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: Convert to using %pOFn instead of device_node.name" to the asoc tree
by Mark Brown 28 Aug '18

28 Aug '18

The patch ASoC: Convert to using %pOFn instead of device_node.name 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 5d585e1e756838d91144c3173323b96f5aa12874 Mon Sep 17 00:00:00 2001 From: Rob Herring <robh(a)kernel.org> Date: Tue, 28 Aug 2018 10:44:28 -0500 Subject: [PATCH] ASoC: Convert to using %pOFn instead of device_node.name In preparation to remove the node name pointer from struct device_node, convert printf users to use the %pOFn format specifier. Signed-off-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_utils.c | 4 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/stm/stm32_sai.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 7 +++---- 5 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c1d1d06783e5..57b484768a58 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -807,7 +807,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); + snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b52223..9981668ab590 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -57,8 +57,8 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, of_node_put(dma_channel_np); return ret; } - snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", - (unsigned long long) res.start, dma_channel_np->name); + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%pOFn", + (unsigned long long) res.start, dma_channel_np); iprop = of_get_property(dma_channel_np, "cell-index", NULL); if (!iprop) { diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 2914ba0d965b..b76a5f4f1785 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -478,7 +478,7 @@ static int axg_card_set_be_link(struct snd_soc_card *card, ret = axg_card_set_link_name(card, link, "be"); if (ret) - dev_err(card->dev, "error setting %s link name\n", np->name); + dev_err(card->dev, "error setting %pOFn link name\n", np); return ret; } diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index f22654253c43..d597eba61992 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -104,7 +104,7 @@ static int stm32_sai_set_sync(struct stm32_sai_data *sai_client, if (!pdev) { dev_err(&sai_client->pdev->dev, - "Device not found for node %s\n", np_provider->name); + "Device not found for node %pOFn\n", np_provider); return -ENODEV; } diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 06fba9650ac4..56a227e0bd71 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1124,16 +1124,15 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->sync = SAI_SYNC_NONE; if (args.np) { if (args.np == np) { - dev_err(&pdev->dev, "%s sync own reference\n", - np->name); + dev_err(&pdev->dev, "%pOFn sync own reference\n", np); of_node_put(args.np); return -EINVAL; } sai->np_sync_provider = of_get_parent(args.np); if (!sai->np_sync_provider) { - dev_err(&pdev->dev, "%s parent node not found\n", - np->name); + dev_err(&pdev->dev, "%pOFn parent node not found\n", + np); of_node_put(args.np); return -ENODEV; } -- 2.18.0

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[alsa-devel] [PATCH 1/7] ASoC: dapm: Remove clock framework ifdefs
by Charles Keepax 28 Aug '18

28 Aug '18

The clock code now has stub functions defined in its header files so the ifdefs around clocking code should no longer be necessary. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 461d951917c0..78ab6965af55 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1320,14 +1320,13 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, soc_dapm_async_complete(w->dapm); -#ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_prepare_enable(w->clk); } else { clk_disable_unprepare(w->clk); return 0; } -#endif + return 0; } EXPORT_SYMBOL_GPL(dapm_clock_event); @@ -3498,7 +3497,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, } break; case snd_soc_dapm_clock_supply: -#ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); @@ -3508,9 +3506,6 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } -#else - return NULL; -#endif break; default: break; -- 2.11.0

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