December 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH v2 1/7][RESEND] ARM: OMAP1: ams-delta: register latch dependent devices later
by Tony Lindgren 20 Dec '11

20 Dec '11

* Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> [111219 14:57]: > Resending because of a typo in the Cc: list, sorry. > > 8<-------------------------- > > In preparation to converting Amstrad Delta on-board latches to > basic_mmio_gpio devices, registration of platform devices which depend > on latches and will require initialization of their GPIO pins first, > should be moved out of .machine_init down to late_initcall level, as the > gpio-generic driver is not available until device_initcall time. The > latch reset operation, which will be replaced with GPIO initialization, > must also be moved to late_initcall for the same reason. > > Since there was already another, separate arch_initcall function for > setting up one of those latch dependent devices, the on-board modem > device, reuse that function, i.e., rename it to a name that matches the > new purpose, extend with other device setup relocated from > .machine_init, and move down to the late_initcall level. > > While being at it, add missing gpio_free() in case the modem platform > device registration fails. > > In addition, defer registration of the Amstrad Delta ASoC and serio > devices, done from their device driver files, until late_initcall time, > as those drivers will depend on their GPIO pins already requested from > the board late_init() function until updated to register their GPIO pins > themselves. > > Thanks to Tony Lindgren <tony(a)atomide.com> who suggested this approach > instead of shifting up the gpio-generic driver initialization. ... > --- a/drivers/input/serio/ams_delta_serio.c > +++ b/drivers/input/serio/ams_delta_serio.c > @@ -165,6 +165,9 @@ serio: > kfree(ams_delta_serio); > return err; > } > +#ifndef MODULE > +late_initcall(ams_delta_serio_init); > +#else > module_init(ams_delta_serio_init); > > static void __exit ams_delta_serio_exit(void) > @@ -175,3 +178,4 @@ static void __exit ams_delta_serio_exit(void) > gpio_free(AMS_DELTA_GPIO_PIN_KEYBRD_DATA); > } > module_exit(ams_delta_serio_exit); > +#endif > diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c > index ccb8a6a..1764a3b 100644 > --- a/sound/soc/omap/ams-delta.c > +++ b/sound/soc/omap/ams-delta.c > @@ -636,6 +636,9 @@ err: > platform_device_put(ams_delta_audio_platform_device); > return ret; > } > +#ifndef MODULE > +late_initcall(ams_delta_module_init); > +#else > module_init(ams_delta_module_init); > > static void __exit ams_delta_module_exit(void) > @@ -657,6 +660,7 @@ static void __exit ams_delta_module_exit(void) > platform_device_unregister(ams_delta_audio_platform_device); > } > module_exit(ams_delta_module_exit); > +#endif > > MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>"); > MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); This looks a bit odd.. I think these drivers should be converted to a proper platform_driver so you can get rid of the machine_is_ams_delta check in the init. Then you can rely on the probe to match the device. To deal with the init order issues, you can pass a set_power function pointer in platform_data that the driver can use. Or set up a fixed regulator for it. Regards, Tony

1 0

[alsa-devel] [PATCH v3] ASoC: Tegra machine ASoC driver for boards using ALC5332 codec
by Leon Romanovsky 20 Dec '11

20 Dec '11

At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> -- Changes since v2: * Remove unsupported frequencies (more then 48Khz) - Suggested by Stephen Warren * Remove GPIO related code, it will be added later by platform_data patchset. Changes since v1: * Use module_platform_driver. * Move controls, dapm_widgets and dapm_routes to snd_soc_card structure. * Set dai_fmt in snd_soc_dai_link structure instead probe function. * Remove check board in probe function * Fix warning about unused codec_dai. * Use devm_kzalloc to manage memory allocation. --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 233 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_alc5632.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd..ce1b773 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3..8e584b8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 0000000..89697c6 --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,213 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Marc Dietrich <marvin24(a)gmx.de> +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 as +* published by the Free Software Foundation. +*/ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 512 * srate; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, + .controls = tegra_alc5632_controls, + .num_controls = ARRAY_SIZE(tegra_alc5632_controls), + .dapm_widgets = tegra_alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), + .dapm_routes = tegra_alc5632_audio_map, + .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), + .fully_routed = true, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + alc5632 = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + return ret; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + tegra_asoc_utils_fini(&alc5632->util_data); + return ret; + } + + return 0; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&alc5632->util_data); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; +module_platform_driver(tegra_alc5632_driver); + +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.4

5 17

[alsa-devel] [PATCH] ASoC: Convert cs42l51 to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 17 ++++++++--------- 1 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 528510b..ffce9f2 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -511,7 +511,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret, reg; ret = cs42l51_fill_cache(codec); @@ -539,20 +538,20 @@ static int cs42l51_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - snd_soc_add_controls(codec, cs42l51_snd_controls, - ARRAY_SIZE(cs42l51_snd_controls)); - snd_soc_dapm_new_controls(dapm, cs42l51_dapm_widgets, - ARRAY_SIZE(cs42l51_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, cs42l51_routes, - ARRAY_SIZE(cs42l51_routes)); - return 0; } static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { - .probe = cs42l51_probe, + .probe = cs42l51_probe, .reg_cache_size = CS42L51_NUMREGS + 1, .reg_word_size = sizeof(u8), + + .controls = cs42l51_snd_controls, + .num_controls = ARRAY_SIZE(cs42l51_snd_controls), + .dapm_widgets = cs42l51_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l51_dapm_widgets), + .dapm_routes = cs42l51_routes, + .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; static int cs42l51_i2c_probe(struct i2c_client *i2c_client, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Convert 88pm860x-codec to table based DAPM and control init
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/88pm860x-codec.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 2d39123..99ca53c 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -861,7 +861,7 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route pm860x_dapm_routes[] = { /* supply */ {"Left DAC", NULL, "VCODEC"}, {"Right DAC", NULL, "VCODEC"}, @@ -1361,7 +1361,6 @@ EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect); static int pm860x_probe(struct snd_soc_codec *codec) { struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int i, ret; pm860x->codec = codec; @@ -1388,11 +1387,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) goto out; } - snd_soc_add_controls(codec, pm860x_snd_controls, - ARRAY_SIZE(pm860x_snd_controls)); - snd_soc_dapm_new_controls(dapm, pm860x_dapm_widgets, - ARRAY_SIZE(pm860x_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); return 0; out: @@ -1420,6 +1414,13 @@ static struct snd_soc_codec_driver soc_codec_dev_pm860x = { .reg_cache_size = REG_CACHE_SIZE, .reg_word_size = sizeof(u8), .set_bias_level = pm860x_set_bias_level, + + .controls = pm860x_snd_controls, + .num_controls = ARRAY_SIZE(pm860x_snd_controls), + .dapm_widgets = pm860x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pm860x_dapm_widgets), + .dapm_routes = pm860x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pm860x_dapm_routes), }; static int __devinit pm860x_codec_probe(struct platform_device *pdev) -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Use dai_fmt in imote2 machine driver
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/imote2.c | 16 ++-------------- 1 files changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 97d3aec..dc905ae 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -30,20 +30,6 @@ static int imote2_asoc_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* CPU should be clock master */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -67,6 +53,8 @@ static struct snd_soc_dai_link imote2_dai = { .codec_dai_name = "wm8940-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm8940-codec.0-0034", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &imote2_asoc_ops, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Use dai_fmt in hx4700 machine driver
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/hx4700.c | 16 ++-------------- 1 files changed, 2 insertions(+), 14 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 03ef9f3..8260207 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -65,20 +65,6 @@ static int hx4700_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the I2S system clock as output */ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_OUT); @@ -175,6 +161,8 @@ static struct snd_soc_dai_link hx4700_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "ak4641.0-0012", .init = hx4700_ak4641_init, + .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &hx4700_ops, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: Convert ak4104 to devm_kzalloc()
by Axel Lin 20 Dec '11

20 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4104.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 152420c..d27b5e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -261,7 +261,8 @@ static int ak4104_spi_probe(struct spi_device *spi) if (ret < 0) return ret; - ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + ak4104 = devm_kzalloc(&spi->dev, sizeof(struct ak4104_private), + GFP_KERNEL); if (ak4104 == NULL) return -ENOMEM; @@ -271,15 +272,12 @@ static int ak4104_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_device_ak4104, &ak4104_dai, 1); - if (ret < 0) - kfree(ak4104); return ret; } static int __devexit ak4104_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } -- 1.7.5.4

1 4

Re: [alsa-devel] [PATCH 5/5] ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
by Mark Brown 20 Dec '11

20 Dec '11

On Wed, Dec 14, 2011 at 03:47:46PM +0800, Shawn Guo wrote: > The commit 49920bc (dmaengine: add new enum dma_transfer_direction) > changes the type of parameter 'direction' of device_prep_dma_cyclic > from dma_data_direction to dma_transfer_direction. Applied, thanks. With patches like this you really need to supply more context - knowing which release the commit you're talking about appears in is useful for exammple, as is having some hint as to what the rest of the series is (does this patch depend on any of the earlier patches, and if not why is it part of a series).

2 1

[alsa-devel] [PATCH v4 1/3] ASoC: Add utility to set a card's name from device tree
by Stephen Warren 20 Dec '11

20 Dec '11

Implement snd_soc_of_parse_card_name(), a utility function that sets a card's name from device tree. The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This series is based on my previously posted patch "ASoC: Allow DAI links to be specified using device tree nodes" v4. Patches 1 & 2 at most rely on that for context. Patch 3 relies on it for functionality. v2: New patch implementing new functionality v3: Name this embedded-audio-complex instead of soc-audio-complex. Change property name from user-visible-name to model. Remove compatible property from binding. v4: Convert this to a utility function that cards can call if desired. Modify snd_soc_register_card() not to call this automatically. Allow cards to specify the name of the device tree property that stores that card's name. Don't specify the property as a generic binding; remove embedded-audio-complex.txt include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 25 +++++++++++++++++++++++++ 2 files changed, 28 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 40c256e..6e51a3f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -966,6 +966,9 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) int snd_soc_util_init(void); void snd_soc_util_exit(void); +int snd_soc_of_parse_card_name(struct snd_soc_card *card, + const char *propname); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83aa82e..4e8d762 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -32,6 +32,7 @@ #include <linux/platform_device.h> #include <linux/ctype.h> #include <linux/slab.h> +#include <linux/of.h> #include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/jack.h> @@ -3367,6 +3368,30 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); +/* Retrieve a card's name from device tree */ +int snd_soc_of_parse_card_name(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + int ret; + + ret = of_property_read_string_index(np, propname, 0, &card->name); + /* + * EINVAL means the property does not exist. This is fine providing + * card->name was previously set, which is checked later in + * snd_soc_register_card. + */ + if (ret < 0 && ret != -EINVAL) { + dev_err(card->dev, + "Property '%s' could not be read: %d\n", + propname, ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls
by Axel Lin 20 Dec '11

20 Dec '11

The module desciption says this is ASoC alc5621/2/3 driver. Make the naming consistent with the reset of the code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index da97f02..6a9b621 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -99,7 +99,7 @@ static const unsigned int boost_tlv[] = { }; static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); -static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -110,7 +110,7 @@ static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { ALC5623_HP_OUT_VOL, 15, 7, 1, 1), }; -static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { +static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = { SOC_DOUBLE_TLV("Speaker Playback Volume", ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), SOC_DOUBLE("Speaker Playback Switch", @@ -925,12 +925,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) switch (alc5623->id) { case 0x21: - snd_soc_add_controls(codec, rt5621_vol_snd_controls, - ARRAY_SIZE(rt5621_vol_snd_controls)); + snd_soc_add_controls(codec, alc5621_vol_snd_controls, + ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: - snd_soc_add_controls(codec, rt5622_vol_snd_controls, - ARRAY_SIZE(rt5622_vol_snd_controls)); + snd_soc_add_controls(codec, alc5622_vol_snd_controls, + ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: snd_soc_add_controls(codec, alc5623_vol_snd_controls, -- 1.7.5.4

2 1