June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v5 0/3] Add ES8328 audio codec
by Sean Cross 21 Jun '14

21 Jun '14

This patchset adds support for the Everest Semi ES8328 audio codec. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks. This requires support for DT clock reparenting in order to get the codec its 22.5792 MHz supply frequency. Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 24 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 63 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 694 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 313 ++++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 263 ++++++++ 12 files changed, 1502 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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Re: [alsa-devel] [PATCH] ASoC: Fix SOC_DOUBLE_R_SX_TLV volume mixer arguments
by Brian Austin 21 Jun '14

21 Jun '14

On Fri, 20 Jun 2014, Ryan Harvey wrote: > Remove unnecessary bit shifts. Correct min value to match datasheet. #Steps equals number of steps between min and max. > > This patch depends on 8f0367b53d887726b1f4b0b39e0f09b7cca2f79f > ASoC: cs42l56: Fix stereo channel register assignment for Headphone and > LineOut volume mixers. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- Acked-by: Brian Austin <brian.austin(a)cirrus.com> Thanks,

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[alsa-devel] [PATCH] ASoC: wm5110: Power both channels for differential mono output
by Charles Keepax 21 Jun '14

21 Jun '14

On the wm5110 CODEC both the left and right channel must be powered when an output is being used as a mono output, although no audio is routed to the right output channel. This patch adds additional DAPM routes to link the right channel to the left in the case where an output is marked as mono. Audio must always be brought in on the left channel for mono operation. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 1 + 3 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 41b56ee..d441d80 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +static const struct snd_soc_dapm_route arizona_mono_routes[] = { + { "OUT1R", NULL, "OUT1L" }, + { "OUT2R", NULL, "OUT2L" }, + { "OUT3R", NULL, "OUT3L" }, + { "OUT4R", NULL, "OUT4L" }, + { "OUT5R", NULL, "OUT5L" }, + { "OUT6R", NULL, "OUT6L" }, +}; + +int arizona_init_mono(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + if (arizona->pdata.out_mono[i]) + snd_soc_dapm_add_routes(&codec->dapm, + &arizona_mono_routes[i], 1); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_mono); + int arizona_init_gpio(struct snd_soc_codec *codec) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 05ae17f..942cfb1 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); +extern int arizona_init_mono(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e5fcb5..62ef544 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_spk(codec); arizona_init_gpio(codec); + arizona_init_mono(codec); ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) -- 1.7.2.5

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[alsa-devel] [PATCH 1/4] ASoC: rt5677: Replace the string "micbias1" to "MICBIAS1"
by Oder Chiou 21 Jun '14

21 Jun '14

The patch replaces the string "micbias1" to "MICBIAS1". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 833231e..75268a7 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1685,7 +1685,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* Input Side */ /* micbias */ - SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT, 0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU), /* Input Lines */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/3] ASoC: rt5640: Remove unneeded goto in rt5640_i2c_probe
by Axel Lin 21 Jun '14

21 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5640.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index de80e89..6bc6efd 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); - if (ret < 0) - goto err; - - return 0; -err: - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); } static int rt5640_i2c_remove(struct i2c_client *i2c) -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: pcm: fix dpcm_path_put in dpcm runtime update
by Qiao Zhou 21 Jun '14

21 Jun '14

we need to release dapm widget list after dpcm_path_get in soc_dpcm_runtime_update. otherwise, there will be potential memory leak. add dpcm_path_put to fix it. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a391de0..3bb8b80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2072,6 +2072,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); } + dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ if (!fe->cpu_dai->driver->capture.channels_min) -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] Couple of fixes to davinci-mcasp driver
by Jyri Sarha 21 Jun '14

21 Jun '14

These changes target to sii9022 HDMI audio support, but are independent fixes. Jyri Sarha (2): ASoC: davinci-mcasp: Add dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC ASoC: davinci-mcasp: Allow best effort in selecting BCLK divider sound/soc/davinci/Kconfig | 1 + sound/soc/davinci/davinci-mcasp.c | 22 +++++++++++++++++----- 2 files changed, 18 insertions(+), 5 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: cs42l56: Fix stereo channel register assignment for Headphone and LineOut volume mixers
by Brian Austin 20 Jun '14

20 Jun '14

On Fri, 20 Jun 2014, Ryan Harvey wrote: > Stereo Headphone and LineOut volume mixers are now attached to HPA+HPB, LOA+LOB. > > Signed-off-by: Ryan Harvey <ryan.harvey(a)cirrus.com> > --- Acked-by: Brian Austin <brian.austin(a)cirrus.com>

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[alsa-devel] [PATCH Resend 1/1] ASoC: mc13783: Add missing of_node_put
by Sachin Kamat 20 Jun '14

20 Jun '14

of_get_child_by_name expects of_node_put be called when done. Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> --- sound/soc/codecs/mc13783.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ARM: dts: imx6: remove wrong spdif rxtx2 clock
by Anssi Hannula 20 Jun '14

20 Jun '14

The rxtx2 clock of i.MX6 S/PDIF is currently set to "asrc" clock. However, according to SoC documentation, rxtx2 is connected to ASRC_EXT_CLK, a different external clock. Testing on Cubox-i system seems to confirm that: when fsl_spdif is forced to select rxtx2 as input clock, audio playback fails with an I/O error. Set rxtx2 to the dummy clock by default to prevent fsl_spdif from selecting it. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: Nicolin Chen <Guangyu.Chen(a)freescale.com> Cc: Mark Brown <broonie(a)kernel.org> --- This one seems clear enough. arch/arm/boot/dts/imx6qdl.dtsi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/boot/dts/imx6qdl.dtsi b/arch/arm/boot/dts/imx6qdl.dtsi index ce0599134a69..3aef9a2f5fda 100644 --- a/arch/arm/boot/dts/imx6qdl.dtsi +++ b/arch/arm/boot/dts/imx6qdl.dtsi @@ -181,7 +181,7 @@ <&sdma 15 18 0>; dma-names = "rx", "tx"; clocks = <&clks 197>, <&clks 3>, - <&clks 197>, <&clks 107>, + <&clks 197>, <&clks 0>, <&clks 0>, <&clks 118>, <&clks 0>, <&clks 139>, <&clks 0>; -- 1.8.4.5

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