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

[alsa-devel] [PATCH] ASoC: Blackfin: fix DAI/SPORT config dependency issues
by Mike Frysinger 13 Jan '11

13 Jan '11

From: Barry Song <barry.song(a)analog.com> While I2S/TDM/AC97 DAI is built-in, others are compiled as modules, SND_BF5XX_SOC_SPORT will be module, then DAI can't get some symbols. Except that, SND_BF5XX_AC97 depends on SND_BF5XX_SOC_AC97 too. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/Kconfig | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 3abeedd..ae40359 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,6 +1,7 @@ config SND_BF5XX_I2S tristate "SoC I2S Audio for the ADI BF5xx chip" depends on BLACKFIN + select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -35,6 +36,7 @@ config SND_BFIN_AD73311_SE config SND_BF5XX_TDM tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" depends on (BLACKFIN && SND_SOC) + select SND_BF5XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in TDM @@ -61,6 +63,10 @@ config SND_BF5XX_SOC_AD193X config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_AC97 help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in slot 16 @@ -122,17 +128,12 @@ config SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_I2S tristate - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_TDM tristate - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SOC_AC97 tristate - select AC97_BUS - select SND_SOC_AC97_BUS - select SND_BF5XX_SOC_SPORT config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" -- 1.7.4.rc1

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[alsa-devel] [PATCH] ASoC: Blackfin AC97: fix build error after multi-component update
by Mike Frysinger 13 Jan '11

13 Jan '11

We need to tweak how we query the active capture/playback state after the recent overhauls of common code. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ac97.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c5f856e..ffbac26 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -260,9 +260,9 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) pr_debug("%s : sport %d\n", __func__, dai->id); if (!dai->active) return 0; - if (dai->capture.active) + if (dai->capture_active) sport_rx_stop(sport); - if (dai->playback.active) + if (dai->playback_active) sport_tx_stop(sport); return 0; } -- 1.7.4.rc1

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[alsa-devel] [PATCH v2 1/3] ASoC: soc-cache: Add support for default readable()/volatile() functions
by Dimitris Papastamos 13 Jan '11

13 Jan '11

For common scenarios, device drivers can provide a table of all the registers that are at least either readable/writable/volatile. The idea is that if a register lookup fails, all of its read/write/vol members will be zero and will be treated as default. This also reduces the size of the register access array. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 22 ++++++++++++++++++++++ sound/soc/soc-cache.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d609232..b8acf99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,6 +276,10 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -367,6 +371,22 @@ int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** + * struct snd_soc_reg_access - Describes whether a given register is + * readable, writable or volatile. + * + * @reg: the register number + * @read: whether this register is readable + * @write: whether this register is writable + * @vol: whether this register is volatile + */ +struct snd_soc_reg_access { + u16 reg; + u16 read; + u16 write; + u16 vol; +}; + +/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -515,6 +535,8 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; + short reg_access_size; + const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1a36b36..d97a59f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1603,3 +1603,52 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); + +static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int min, max, index; + + codec_drv = codec->driver; + min = 0; + max = codec_drv->reg_access_size - 1; + do { + index = (min + max) / 2; + if (codec_drv->reg_access_default[index].reg == reg) + return index; + if (codec_drv->reg_access_default[index].reg < reg) + min = index + 1; + else + max = index; + } while (min <= max); + return -1; +} + +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].vol; +} +EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); + +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].read; +} +EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); -- 1.7.3.5

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[alsa-devel] [PATCH] ASoC: tegra: s/IS_ERR_OR_NULL/IS_ERR/ for clk_get_sys
by Stephen Warren 13 Jan '11

13 Jan '11

A recent discussion on linux-arm-kernel noted that the value returned by clk_get_sys is an opaque token, and not strictly a pointer; it is meaningful only to the clock API, clients should not dereference the value, and the clock API must accept any non-IS_ERR value it returned. Hence, only IS_ERR is appropriate to interpret the result, not IS_ERR_OR_NULL. I checked that clk_get_sys in both ASoC's for-next and Tegra's for-next do behave as described; NULL is not returned in the case of error. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_asoc_utils.c | 20 +++++++++----------- sound/soc/tegra/tegra_i2s.c | 2 +- 2 files changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 711ab7f..cfe2ea8 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -113,35 +113,33 @@ int tegra_asoc_utils_init(void) int ret; clk_pll_a = clk_get_sys(NULL, "pll_a"); - if (IS_ERR_OR_NULL(clk_pll_a)) { + if (IS_ERR(clk_pll_a)) { pr_err(PREFIX "Can't retrieve clk pll_a\n"); ret = PTR_ERR(clk_pll_a); goto err; } clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); - if (IS_ERR_OR_NULL(clk_pll_a_out0)) { + if (IS_ERR(clk_pll_a_out0)) { pr_err(PREFIX "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(clk_pll_a_out0); - goto err; + goto err_put_pll_a; } clk_cdev1 = clk_get_sys(NULL, "cdev1"); - if (IS_ERR_OR_NULL(clk_cdev1)) { + if (IS_ERR(clk_cdev1)) { pr_err(PREFIX "Can't retrieve clk cdev1\n"); ret = PTR_ERR(clk_cdev1); - goto err; + goto err_put_pll_a_out0; } return 0; +err_put_pll_a_out0: + clk_put(clk_pll_a_out0); +err_put_pll_a: + clk_put(clk_pll_a); err: - if (!IS_ERR_OR_NULL(clk_cdev1)) - clk_put(clk_cdev1); - if (!IS_ERR_OR_NULL(clk_pll_a_out0)) - clk_put(clk_pll_a_out0); - if (!IS_ERR_OR_NULL(clk_pll_a)) - clk_put(clk_pll_a); return ret; } diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 9b7a22a..75d3bfe 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -385,7 +385,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id); i2s->clk_i2s = clk_get_sys(clk_name, NULL); - if (IS_ERR_OR_NULL(i2s->clk_i2s)) { + if (IS_ERR(i2s->clk_i2s)) { pr_err("Can't retrieve i2s clock\n"); ret = PTR_ERR(i2s->clk_i2s); goto err_free; -- 1.7.0.4

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[alsa-devel] [RFC] [ALSA] ASoC documentation updates
by Seungwhan Youn 13 Jan '11

13 Jan '11

This patch is only for RFC purpose of ASoC documentation updates which match with current ASoC codes with documents. Mostly modify features are modified to be sync with changes after multi-component patches. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- To Mr. Liam Girdwood and Mark Brown I think this modification is not enough to cover current ASoC features, and I hope that it will be a good time to introduce recently changes, e.g. multi-component model, new linkage system of ASoC DAIs, and so on. But if you have some other idea, or other reason to delay this, please ignore this RFC. And also if there are something that I can help, please let me know. Thanks, Claude --- Documentation/sound/alsa/soc/codec.txt | 45 +++++++++++++--------------- Documentation/sound/alsa/soc/machine.txt | 38 ++++++------------------ Documentation/sound/alsa/soc/platform.txt | 12 ++++++- 3 files changed, 40 insertions(+), 55 deletions(-) diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 37ba3a7..bce23a4 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -27,42 +27,38 @@ ASoC Codec driver breakdown 1 - Codec DAI and PCM configuration ----------------------------------- -Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +Each codec driver must have a struct snd_soc_dai_driver to define its DAI and PCM capabilities and operations. This struct is exported so that it can be registered with the core by your machine driver. e.g. -struct snd_soc_codec_dai wm8731_dai = { - .name = "WM8731", - /* playback capabilities */ +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + +struct snd_soc_dai_driver wm8731_dai = { + .name = "wm8731-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - /* capture capabilities */ .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - /* pcm operations - see section 4 below */ - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - }, - /* DAI operations - see DAI.txt */ - .dai_ops = { - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, + .symmetric_rates = 1, }; -EXPORT_SYMBOL_GPL(wm8731_dai); 2 - Codec control IO @@ -186,13 +182,14 @@ when the mute is applied or freed. i.e. -static int wm8974_mute(struct snd_soc_codec *codec, - struct snd_soc_codec_dai *dai, int mute) +static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; - if(mute) - wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + + if (mute) + snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); else - wm8974_write(codec, WM8974_DAC, mute_reg); + snd_soc_write(codec, WM8974_DAC, mute_reg); return 0; } diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 2524c75..3e2ec9c 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -12,6 +12,8 @@ the following struct:- struct snd_soc_card { char *name; + ... + int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -22,12 +24,13 @@ struct snd_soc_card { int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev); - /* machine stream operations */ - struct snd_soc_ops *ops; + ... /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + + ... }; probe()/remove() @@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs and DMA is suspended and resumed. Optional. -Machine operations ------------------- -The machine specific audio operations can be set here. Again this is optional. - - Machine DAI Configuration ------------------------- The machine DAI configuration glues all the codec and CPU DAIs together. It can @@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai = &pxa_i2s_dai, - .codec_dai = &wm8731_dai, + .cpu_dai_name = "pxa-is2-dai", + .codec_dai_name = "wm8731-hifi", + .platform_name = "pxa-pcm-audio", + .codec_name = "wm8713-codec.0-001a", .init = corgi_wm8731_init, .ops = &corgi_ops, }; @@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = { }; -Machine Audio Subsystem ------------------------ - -The machine soc device glues the platform, machine and codec driver together. -Private data can also be set here. e.g. - -/* corgi audio private data */ -static struct wm8731_setup_data corgi_wm8731_setup = { - .i2c_address = 0x1b, -}; - -/* corgi audio subsystem */ -static struct snd_soc_device corgi_snd_devdata = { - .machine = &snd_soc_corgi, - .platform = &pxa2xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &corgi_wm8731_setup, -}; - - Machine Power Map ----------------- diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index 06d8359..d57efad 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -20,9 +20,10 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -The platform driver exports its DMA functionality via struct snd_soc_platform:- +The platform driver exports its DMA functionality via struct +snd_soc_platform_driver:- -struct snd_soc_platform { +struct snd_soc_platform_driver { char *name; int (*probe)(struct platform_device *pdev); @@ -34,6 +35,13 @@ struct snd_soc_platform { int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); + /* + * For platform caused delay reporting. + * Optional. + */ + snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, + struct snd_soc_dai *); + /* platform stream ops */ struct snd_pcm_ops *pcm_ops; }; -- 1.6.2.5

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[alsa-devel] [PATCH] ASoC: Blackfin TDM: use external frame syncs
by Mike Frysinger 13 Jan '11

13 Jan '11

From: Barry Song <barry.song(a)analog.com> We don't want to use internal frame syncs otherwise we sometimes get out of sync, so don't enable them when setting up the SPORT. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-tdm.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index b2cf239..5515ac9 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -235,13 +235,13 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) ret = -EBUSY; } - ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0); + ret = sport_config_rx(sport, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; } - ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0); + ret = sport_config_tx(sport, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; @@ -303,14 +303,14 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) goto sport_config_err; } - ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0); + ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; goto sport_config_err; } - ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0); + ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- 1.7.4.rc1

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[alsa-devel] [PATCH 1/3] ASoC: soc-cache: Add support for default readable()/volatile() functions
by Dimitris Papastamos 13 Jan '11

13 Jan '11

For common scenarios, device drivers can provide a table of all the registers that are at least either readable/writable/volatile. The idea is that if a register lookup fails, all of its read/write/vol members will be zero and will be treated as default. This also reduces the size of the register access array. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 22 ++++++++++++++++++++++ sound/soc/soc-cache.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 71 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index d609232..b8acf99 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,6 +276,10 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg); +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -367,6 +371,22 @@ int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** + * struct snd_soc_reg_access - Describes whether a given register is + * readable, writable or volatile. + * + * @reg: the register number + * @read: whether this register is readable + * @write: whether this register is writable + * @vol: whether this register is volatile + */ +struct snd_soc_reg_access { + u16 reg; + u16 read; + u16 write; + u16 vol; +}; + +/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -515,6 +535,8 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; + short reg_access_size; + const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1a36b36..d97a59f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1603,3 +1603,52 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); + +static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int min, max, index; + + codec_drv = codec->driver; + min = 0; + max = codec_drv->reg_access_size - 1; + do { + index = (min + max) / 2; + if (codec_drv->reg_access_default[index].reg == reg) + return index; + if (codec_drv->reg_access_default[index].reg < reg) + min = index + 1; + else + max = index; + } while (min <= max); + return -1; +} + +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].vol; +} +EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); + +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].read; +} +EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); -- 1.7.3.5

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[alsa-devel] registering multiple cards
by Koul, Vinod 13 Jan '11

13 Jan '11

Hi, Currently snd_soc_register_card is internal soc-core API and is called from soc-probe. And probe always assumes it is a single device, so how do we add up two cards? I saw a note on this that it will exposed so that drivers can register. What is that plan to do that, and meanwhile what can be done to solve this? ~Vinod

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[alsa-devel] [PATCH] powerpc/85xx: fix compatible properties of the P1022DS DMA nodes used for audio
by Timur Tabi 13 Jan '11

13 Jan '11

In order to prevent the fsl_dma driver from claiming the DMA channels that the P1022DS audio driver needs, the compatible properties for those nodes must say "fsl,ssi-dma-channel" instead of "fsl,eloplus-dma-channel". Signed-off-by: Timur Tabi <timur(a)freescale.com> --- Kumar, please pick up this patch for 2.6.37. arch/powerpc/boot/dts/p1022ds.dts | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/powerpc/boot/dts/p1022ds.dts b/arch/powerpc/boot/dts/p1022ds.dts index 2bbecbb..69422eb 100644 --- a/arch/powerpc/boot/dts/p1022ds.dts +++ b/arch/powerpc/boot/dts/p1022ds.dts @@ -291,13 +291,13 @@ ranges = <0x0 0xc100 0x200>; cell-index = <1>; dma00: dma-channel@0 { - compatible = "fsl,eloplus-dma-channel"; + compatible = "fsl,ssi-dma-channel"; reg = <0x0 0x80>; cell-index = <0>; interrupts = <76 2>; }; dma01: dma-channel@80 { - compatible = "fsl,eloplus-dma-channel"; + compatible = "fsl,ssi-dma-channel"; reg = <0x80 0x80>; cell-index = <1>; interrupts = <77 2>; -- 1.7.2.3

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[alsa-devel] [PATCH 0/2] ASoC: sh: fsi-codecs: add/modify FSI selection
by Kuninori Morimoto 13 Jan '11

13 Jan '11

Dear Mark, Liam Kuninori Morimoto (2): ASoC: sh: fsi-ak4642: fixup platform device id ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection 1st patch fixup platform device id. 2nd patch add FSI PortA/B selection to FSI-HDMI These patches are tested on Ecovec, AP4EVB, Mackerel board, and Paul's rmobile-latest + Mark's for-2.6.38 branch. Best regards -- Kuninori Morimoto

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