From 1ef6ab75c7deef931d6308af282ed7d8e480e77f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:55 +0200 Subject: [ALSA] ASoC: Make CPU and codec DAI operations have same type The CPU and codec DAI operations differ only in the presence of the digital mute operation for the codec so they may as well be the same type. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 30 ++++++------------------------ 1 file changed, 6 insertions(+), 24 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index d3c8c033dff8..73accbcfbd2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -272,9 +272,9 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* ASoC codec DAI ops */ -struct snd_soc_codec_ops { - /* codec DAI clocking configuration */ +/* ASoC DAI ops */ +struct snd_soc_dai_ops { + /* DAI clocking configuration */ int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, int clk_id, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec_dai *codec_dai, @@ -282,7 +282,7 @@ struct snd_soc_codec_ops { int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, int div_id, int div); - /* CPU DAI format configuration */ + /* DAI format configuration */ int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, @@ -293,24 +293,6 @@ struct snd_soc_codec_ops { int (*digital_mute)(struct snd_soc_codec_dai *, int mute); }; -/* ASoC cpu DAI ops */ -struct snd_soc_cpu_ops { - /* CPU DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_cpu_dai *cpu_dai, - int clk_id, unsigned int freq, int dir); - int (*set_clkdiv)(struct snd_soc_cpu_dai *cpu_dai, - int div_id, int div); - int (*set_pll)(struct snd_soc_cpu_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - - /* CPU DAI format configuration */ - int (*set_fmt)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_cpu_dai *cpu_dai, - unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_cpu_dai *, int tristate); -}; - /* SoC Codec DAI */ struct snd_soc_codec_dai { char *name; @@ -328,7 +310,7 @@ struct snd_soc_codec_dai { /* ops */ struct snd_soc_ops ops; - struct snd_soc_codec_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI private data */ void *private_data; @@ -352,7 +334,7 @@ struct snd_soc_cpu_dai { /* ops */ struct snd_soc_ops ops; - struct snd_soc_cpu_ops dai_ops; + struct snd_soc_dai_ops dai_ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; -- cgit v1.2.3 From 0be9898adb6f58fee44f0fec0bbc0eac997ea9eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:31:28 +0200 Subject: [ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown Cc: Jarkko Nikula Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 28 +++++++++++++++++++++++---- sound/soc/codecs/tlv320aic3x.c | 26 ++++++++++++------------- sound/soc/codecs/wm8731.c | 28 +++++++++++++-------------- sound/soc/codecs/wm8750.c | 36 +++++++++++++++++----------------- sound/soc/codecs/wm8753.c | 36 +++++++++++++++++----------------- sound/soc/codecs/wm9712.c | 28 +++++++++++++-------------- sound/soc/codecs/wm9713.c | 26 ++++++++++++------------- sound/soc/soc-core.c | 25 ++++++++++++------------ sound/soc/soc-dapm.c | 44 ++++++++++++++++++++---------------------- 10 files changed, 150 insertions(+), 130 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bf4cf0c1d37f..f8223fae5804 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -221,7 +221,8 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); -int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event); +int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index 73accbcfbd2d..bca9538d9e50 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -102,6 +102,24 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +/* + * Bias levels + * + * @ON: Bias is fully on for audio playback and capture operations. + * @PREPARE: Prepare for audio operations. Called before DAPM switching for + * stream start and stop operations. + * @STANDBY: Low power standby state when no playback/capture operations are + * in progress. NOTE: The transition time between STANDBY and ON + * should be as fast as possible and no longer than 10ms. + * @OFF: Power Off. No restrictions on transition times. + */ +enum snd_soc_bias_level { + SND_SOC_BIAS_ON, + SND_SOC_BIAS_PREPARE, + SND_SOC_BIAS_STANDBY, + SND_SOC_BIAS_OFF, +}; + /* * Digital Audio Interface (DAI) types */ @@ -356,7 +374,8 @@ struct snd_soc_codec { struct mutex mutex; /* callbacks */ - int (*dapm_event)(struct snd_soc_codec *codec, int event); + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); /* runtime */ struct snd_card *card; @@ -378,8 +397,8 @@ struct snd_soc_codec { /* dapm */ struct list_head dapm_widgets; struct list_head dapm_paths; - unsigned int dapm_state; - unsigned int suspend_dapm_state; + enum snd_soc_bias_level bias_level; + enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; /* codec DAI's */ @@ -449,7 +468,8 @@ struct snd_soc_machine { int (*resume_post)(struct platform_device *pdev); /* callbacks */ - int (*dapm_event)(struct snd_soc_machine *, int event); + int (*set_bias_level)(struct snd_soc_machine *, + enum snd_soc_bias_level level); /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb8365ac0c02..dc8a38d9e53a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -847,13 +847,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) +static int aic3x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = codec->private_data; u8 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: + switch (level) { + case SND_SOC_BIAS_ON: /* all power is driven by DAPM system */ if (aic3x->master) { /* enable pll */ @@ -862,10 +863,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg | PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D1: - case SNDRV_CTL_POWER_D2: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: + case SND_SOC_BIAS_STANDBY: /* * all power is driven by DAPM system, * so output power is safe if bypass was set @@ -877,7 +877,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) reg & ~PLL_ENABLE); } break; - case SNDRV_CTL_POWER_D3cold: + case SND_SOC_BIAS_OFF: /* force all power off */ reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL); aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON); @@ -913,7 +913,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event) } break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -979,7 +979,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -999,7 +999,7 @@ static int aic3x_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - aic3x_dapm_event(codec, codec->suspend_dapm_state); + aic3x_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -1018,7 +1018,7 @@ static int aic3x_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; - codec->dapm_event = aic3x_dapm_event; + codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(aic3x_reg); @@ -1100,7 +1100,7 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); /* off, with power on */ - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* setup GPIO functions */ aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); @@ -1271,7 +1271,7 @@ static int aic3x_remove(struct platform_device *pdev) /* power down chip */ if (codec->control_data) - aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3); + aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0cf9265fca8f..0f28aa4bcccb 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -435,29 +435,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8731_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ wm8731_write(codec, WM8731_PWR, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ wm8731_write(codec, WM8731_PWR, reg | 0x0040); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_write(codec, WM8731_PWR, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -503,7 +503,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -521,8 +521,8 @@ static int wm8731_resume(struct platform_device *pdev) data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); } - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); - wm8731_dapm_event(codec, codec->suspend_dapm_state); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8731_set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -539,7 +539,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8731_read_reg_cache; codec->write = wm8731_write; - codec->dapm_event = wm8731_dapm_event; + codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(wm8731_reg); @@ -557,7 +557,7 @@ static int wm8731_init(struct snd_soc_device *socdev) } /* power on device */ - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); @@ -730,7 +730,7 @@ static int wm8731_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 16cd5d4d5ad9..62423f4493b0 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -686,29 +686,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8750_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8750_write(codec, WM8750_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -748,7 +748,7 @@ static void wm8750_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8750_dapm_event(codec, codec->dapm_state); + wm8750_set_bias_level(codec, codec->bias_level); } static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) @@ -756,7 +756,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -777,12 +777,12 @@ static int wm8750_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8750 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); } @@ -803,7 +803,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8750_read_reg_cache; codec->write = wm8750_write; - codec->dapm_event = wm8750_dapm_event; + codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; codec->reg_cache_size = sizeof(wm8750_reg); @@ -821,8 +821,8 @@ static int wm8750_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -1021,7 +1021,7 @@ static int wm8750_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index fb41826c4c4c..9032b0c07c86 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1274,29 +1274,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) +static int wm8753_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: wm8753_write(codec, WM8753_PWR1, 0x0001); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1500,7 +1500,7 @@ static void wm8753_work(struct work_struct *work) { struct snd_soc_codec *codec = container_of(work, struct snd_soc_codec, delayed_work.work); - wm8753_dapm_event(codec, codec->dapm_state); + wm8753_set_bias_level(codec, codec->bias_level); } static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) @@ -1512,7 +1512,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) if (!codec->card) return 0; - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1537,12 +1537,12 @@ static int wm8753_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* charge wm8753 caps */ - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D0; + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_ON; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); } @@ -1563,7 +1563,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->owner = THIS_MODULE; codec->read = wm8753_read_reg_cache; codec->write = wm8753_write; - codec->dapm_event = wm8753_dapm_event; + codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; codec->reg_cache_size = sizeof(wm8753_reg); @@ -1584,8 +1584,8 @@ static int wm8753_init(struct snd_soc_device *socdev) } /* charge output caps */ - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); - codec->dapm_state = SNDRV_CTL_POWER_D3hot; + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(caps_charge)); @@ -1792,7 +1792,7 @@ static int wm8753_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) - wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 89efe40c7c33..e26cfcf0b4fc 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -571,23 +571,23 @@ struct snd_soc_codec_dai wm9712_dai[] = { }; EXPORT_SYMBOL_GPL(wm9712_dai); -static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9712_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -615,7 +615,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -632,7 +632,7 @@ static int wm9712_soc_resume(struct platform_device *pdev) return ret; } - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret == 0) { /* Sync reg_cache with the hardware after cold reset */ @@ -644,8 +644,8 @@ static int wm9712_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9712_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -679,7 +679,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9712_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9712_dapm_event; + codec->set_bias_level = wm9712_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -703,7 +703,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); - wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm9712_add_controls(codec); wm9712_add_widgets(codec); ret = snd_soc_register_card(socdev); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 9e6b2fd7262b..4863636e9d56 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1094,33 +1094,33 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) } EXPORT_SYMBOL_GPL(wm9713_reset); -static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +static int wm9713_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { u16 reg; - switch (event) { - case SNDRV_CTL_POWER_D0: /* full On */ + switch (level) { + case SND_SOC_BIAS_ON: /* enable thermal shutdown */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; ac97_write(codec, AC97_EXTENDED_MID, reg); break; - case SNDRV_CTL_POWER_D1: /* partial On */ - case SNDRV_CTL_POWER_D2: /* partial On */ + case SND_SOC_BIAS_PREPARE: break; - case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; - case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } - codec->dapm_state = event; + codec->bias_level = level; return 0; } @@ -1157,7 +1157,7 @@ static int wm9713_soc_resume(struct platform_device *pdev) return ret; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ if (wm9713->pll_out) @@ -1173,8 +1173,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) } } - if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm9713_set_bias_level(codec, SND_SOC_BIAS_ON); return ret; } @@ -1213,7 +1213,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) codec->num_dai = ARRAY_SIZE(wm9713_dai); codec->write = ac97_write; codec->read = ac97_read; - codec->dapm_event = wm9713_dapm_event; + codec->set_bias_level = wm9713_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1235,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) goto reset_err; } - wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0318d8abe3e8..a05b3450aee8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -283,12 +283,12 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { - /* power down the codec to D1 if no longer active */ + /* Reduce power if no longer active */ if (codec->active == 0) { dbg("pop wq D1 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); } codec_dai->pop_wait = 0; @@ -296,12 +296,12 @@ static void close_delayed_work(struct work_struct *work) codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_STOP); - /* power down the codec power domain if no longer active */ + /* Fall into standby if no longer active */ if (codec->active == 0) { dbg("pop wq D3 %s %s\n", codec->name, codec_dai->playback.stream_name); - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } } } @@ -361,8 +361,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); if (codec->active == 0 && codec_dai->pop_wait == 0) - snd_soc_dapm_device_event(socdev, - SNDRV_CTL_POWER_D3hot); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); } mutex_unlock(&pcm_mutex); @@ -435,9 +435,10 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } else { /* no delayed work - do we need to power up codec */ - if (codec->dapm_state != SNDRV_CTL_POWER_D0) { + if (codec->bias_level != SND_SOC_BIAS_ON) { - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D1); + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dapm_stream_event(codec, @@ -448,7 +449,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); if (codec_dai->dai_ops.digital_mute) codec_dai->dai_ops.digital_mute(codec_dai, 0); @@ -658,7 +659,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* close any waiting streams and save state */ run_delayed_work(&socdev->delayed_work); - codec->suspend_dapm_state = codec->dapm_state; + codec->suspend_bias_level = codec->bias_level; for(i = 0; i < codec->num_dai; i++) { char *stream = codec->dai[i].playback.stream_name; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8a3192bcee78..728f3ac2f304 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -763,21 +763,18 @@ static ssize_t dapm_widget_show(struct device *dev, } } - switch(codec->dapm_state){ - case SNDRV_CTL_POWER_D0: - state = "D0"; + switch (codec->bias_level) { + case SND_SOC_BIAS_ON: + state = "On"; break; - case SNDRV_CTL_POWER_D1: - state = "D1"; + case SND_SOC_BIAS_PREPARE: + state = "Prepare"; break; - case SNDRV_CTL_POWER_D2: - state = "D2"; + case SND_SOC_BIAS_STANDBY: + state = "Standby"; break; - case SNDRV_CTL_POWER_D3hot: - state = "D3hot"; - break; - case SNDRV_CTL_POWER_D3cold: - state = "D3cold"; + case SND_SOC_BIAS_OFF: + state = "Off"; break; } count += sprintf(buf + count, "PM State: %s\n", state); @@ -1358,27 +1355,28 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** - * snd_soc_dapm_device_event - send a device event to the dapm core + * snd_soc_dapm_set_bias_level - set the bias level for the system * @socdev: audio device - * @event: device event + * @level: level to configure * - * Sends a device event to the dapm core. The core then makes any - * necessary machine or codec power changes.. + * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event) +int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, + enum snd_soc_bias_level level) { struct snd_soc_codec *codec = socdev->codec; struct snd_soc_machine *machine = socdev->machine; + int ret = 0; - if (machine->dapm_event) - machine->dapm_event(machine, event); - if (codec->dapm_event) - codec->dapm_event(codec, event); - return 0; + if (machine->set_bias_level) + ret = machine->set_bias_level(machine, level); + if (ret == 0 && codec->set_bias_level) + ret = codec->set_bias_level(codec, level); + + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event); /** * snd_soc_dapm_set_endpoint - set audio endpoint status -- cgit v1.2.3 From e13ac2e9b18bde51cf32c69c2209df25791ab3e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 28 May 2008 17:58:05 +0100 Subject: [ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type The SOC_DOUBLE_S8_TLV control type was originally implemented in the UDA1380 driver by Philipp Zabel and was moved into the core by me. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/soc.h | 15 +++++++++++ sound/soc/soc-core.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index bca9538d9e50..9fa2093e74eb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -73,6 +73,15 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } +#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | \ + (((signed char)min) << 24) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -267,6 +276,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /* SoC PCM stream information */ struct snd_soc_pcm_stream { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a3f091e0843a..f594ab888e17 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1589,6 +1589,78 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int max = (signed char)((kcontrol->private_value >> 16) & 0xff); + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + int val = snd_soc_read(codec, reg); + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + unsigned short val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_update_bits(codec, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- cgit v1.2.3 From bdb92876f0a9d2b431199e385732ede89ff0b97d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:10 +0100 Subject: ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove This allows per-DAI initialisation to be done by the CPU DAI drivers. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 6 ++++-- sound/soc/davinci/davinci-i2s.c | 6 ++++-- sound/soc/pxa/pxa2xx-ac97.c | 3 ++- sound/soc/s3c24xx/s3c2412-i2s.c | 3 ++- sound/soc/s3c24xx/s3c2443-ac97.c | 6 ++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 3 ++- sound/soc/soc-core.c | 6 +++--- 7 files changed, 21 insertions(+), 12 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9fa2093e74eb..56d2224c2c07 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -358,8 +358,10 @@ struct snd_soc_cpu_dai { unsigned char type; /* DAI callbacks */ - int (*probe)(struct platform_device *pdev); - void (*remove)(struct platform_device *pdev); + int (*probe)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c421774b33ee..c3b545ccff72 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -295,7 +295,8 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static int davinci_i2s_probe(struct platform_device *pdev) +static int davinci_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; @@ -356,7 +357,8 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev) +static void davinci_i2s_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 97ec2d90547c..cb947956ed1a 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct platform_device *pdev) +static int pxa2xx_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c4a46dd589b3..c463a82dec3a 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void) EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); -static int s3c2412_i2s_probe(struct platform_device *pdev) +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 0eed140dcd9b..533565b61b2f 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -209,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { .dma_size = 4, }; -static int s3c2443_ac97_probe(struct platform_device *pdev) +static int s3c2443_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; u32 ac_glbctrl; @@ -260,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) return ret; } -static void s3c2443_ac97_remove(struct platform_device *pdev) +static void s3c2443_ac97_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4c52f7946d9e..42e96b5ff825 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -377,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void) } EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); -static int s3c24xx_i2s_probe(struct platform_device *pdev) +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f594ab888e17..c96a6184d66e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -768,7 +768,7 @@ static int soc_probe(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev); + ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) goto cpu_dai_err; } @@ -798,7 +798,7 @@ cpu_dai_err: for (i--; i >= 0; i--) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) @@ -827,7 +827,7 @@ static int soc_remove(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); } if (machine->remove) -- cgit v1.2.3 From 10144c09a0d6a62e1d56e25f142743c7a00e5dba Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:13 +0100 Subject: ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 56d2224c2c07..1f5c62181002 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -100,6 +100,15 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } +#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ -- cgit v1.2.3 From 6ed2597883b1b03ca94f62f0cfe908314cba6d6b Mon Sep 17 00:00:00 2001 From: Andy Green Date: Fri, 13 Jun 2008 16:24:05 +0100 Subject: ALSA: ASoC: Don't block system resume On OpenMoko soc-audio resume is taking 700ms of the whole resume time of 1.3s, dominated by writes to the codec over I2C. This patch shunts the resume guts into a workqueue which then is done asynchronously. The "card" is locked using the ALSA power state APIs as suggested by Mark Brown. [Added fix for race with resume to suspend and fixed a couple of nits from checkpatch -- broonie.] Signed-off-by: Andy Green Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 46 +++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 44 insertions(+), 3 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1f5c62181002..340223a8f24c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -510,6 +510,7 @@ struct snd_soc_device { struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; struct delayed_work delayed_work; + struct work_struct deferred_resume_work; void *codec_data; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c96a6184d66e..b931039632c5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -639,6 +639,16 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; int i; + /* Due to the resume being scheduled into a workqueue we could + * suspend before that's finished - wait for it to complete. + */ + snd_power_lock(codec->card); + snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); + snd_power_unlock(codec->card); + + /* we're going to block userspace touching us until resume completes */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); + /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; @@ -691,16 +701,27 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -/* powers up audio subsystem after a suspend */ -static int soc_resume(struct platform_device *pdev) +/* deferred resume work, so resume can complete before we finished + * setting our codec back up, which can be very slow on I2C + */ +static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_device *socdev = container_of(work, + struct snd_soc_device, + deferred_resume_work); struct snd_soc_machine *machine = socdev->machine; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; + struct platform_device *pdev = to_platform_device(socdev->dev); int i; + /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, + * so userspace apps are blocked from touching us + */ + + dev_info(socdev->dev, "starting resume work\n"); + if (machine->resume_pre) machine->resume_pre(pdev); @@ -742,6 +763,22 @@ static int soc_resume(struct platform_device *pdev) if (machine->resume_post) machine->resume_post(pdev); + dev_info(socdev->dev, "resume work completed\n"); + + /* userspace can access us now we are back as we were before */ + snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); +} + +/* powers up audio subsystem after a suspend */ +static int soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + dev_info(socdev->dev, "scheduling resume work\n"); + + if (!schedule_work(&socdev->deferred_resume_work)) + dev_err(socdev->dev, "work item may be lost\n"); + return 0; } @@ -788,6 +825,9 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); + /* deferred resume work */ + INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); + return 0; platform_err: -- cgit v1.2.3 From 3c4b266fe642bcaebe2b95edb56c3f8802924ff9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:17 +0100 Subject: ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 71 +++++++++++++++++----------------------------------- sound/soc/soc-core.c | 50 ++++++++++++++++++------------------ 2 files changed, 48 insertions(+), 73 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 340223a8f24c..778e57e74dc8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -221,8 +221,7 @@ struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; -struct snd_soc_codec_dai; -struct snd_soc_cpu_dai; +struct snd_soc_dai; struct snd_soc_codec; struct snd_soc_machine_config; struct soc_enum; @@ -317,50 +316,24 @@ struct snd_soc_ops { /* ASoC DAI ops */ struct snd_soc_dai_ops { /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, + int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec_dai *codec_dai, + int (*set_pll)(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, - int div_id, int div); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_codec_dai *, int tristate); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* digital mute */ - int (*digital_mute)(struct snd_soc_codec_dai *, int mute); + int (*digital_mute)(struct snd_soc_dai *dai, int mute); }; -/* SoC Codec DAI */ -struct snd_soc_codec_dai { - char *name; - int id; - unsigned char type; - - /* DAI capabilities */ - struct snd_soc_pcm_stream playback; - struct snd_soc_pcm_stream capture; - - /* DAI runtime info */ - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI private data */ - void *private_data; -}; - -/* SoC CPU DAI */ -struct snd_soc_cpu_dai { - +/* SoC DAI (Digital Audio Interface) */ +struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; @@ -368,13 +341,13 @@ struct snd_soc_cpu_dai { /* DAI callbacks */ int (*probe)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); void (*remove)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* ops */ struct snd_soc_ops ops; @@ -386,7 +359,9 @@ struct snd_soc_cpu_dai { /* DAI runtime info */ struct snd_pcm_runtime *runtime; - unsigned char active:1; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; void *dma_data; /* DAI private data */ @@ -428,7 +403,7 @@ struct snd_soc_codec { struct delayed_work delayed_work; /* codec DAI's */ - struct snd_soc_codec_dai *dai; + struct snd_soc_dai *dai; unsigned int num_dai; }; @@ -447,12 +422,12 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, + int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); @@ -466,8 +441,8 @@ struct snd_soc_dai_link { char *stream_name; /* Stream name */ /* DAI */ - struct snd_soc_codec_dai *codec_dai; - struct snd_soc_cpu_dai *cpu_dai; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bdbbc6a980fa..4d626b47b2ff 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -134,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -272,7 +272,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; mutex_lock(&pcm_mutex); @@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; int ret = 0; @@ -489,8 +489,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -559,8 +559,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -594,8 +594,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { @@ -651,7 +651,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); } @@ -664,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) machine->suspend_pre(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) @@ -690,7 +690,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) codec_dev->suspend(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); } @@ -726,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work) machine->resume_pre(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); } @@ -747,13 +747,13 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) @@ -803,7 +803,7 @@ static int soc_probe(struct platform_device *pdev) } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -838,7 +838,7 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -867,7 +867,7 @@ static int soc_remove(struct platform_device *pdev) codec_dev->remove(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -895,8 +895,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = dai_link->codec_dai; + struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; struct snd_pcm *pcm; char new_name[64]; @@ -1211,7 +1211,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; #ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; #endif -- cgit v1.2.3 From 8c6529dbf881303920a415c2d14a500218661949 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:13 +0100 Subject: ALSA: asoc: core - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- include/sound/soc.h | 21 ++++++++ sound/soc/soc-core.c | 140 ++++++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 153 insertions(+), 8 deletions(-) (limited to 'include/sound/soc.h') diff --git a/include/sound/soc.h b/include/sound/soc.h index 778e57e74dc8..1890d87c5204 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + /* *Controls */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d626b47b2ff..83f1190293a8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -434,8 +434,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) else { codec_dai->pop_wait = 0; cancel_delayed_work(&socdev->delayed_work); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } } else { /* no delayed work - do we need to power up codec */ @@ -454,8 +453,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_START); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } else { /* codec already powered - power on widgets */ @@ -467,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + + snd_soc_dai_digital_mute(codec_dai, 0); } } @@ -566,8 +564,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* apply codec digital mute */ - if (!codec->active && codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 1); + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) @@ -1703,6 +1701,132 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @clk_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->dai_ops.set_clkdiv) + return dai->dai_ops.set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + if (dai->dai_ops.set_pll) + return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @fmt: SND_SOC_DAIFMT_ format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->dai_ops.set_fmt) + return dai->dai_ops.set_fmt(dai, fmt); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @mask: DAI specific mask representing used slots. + * @slots: Number of slots in use. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tdm_slot(dai, mask, slots); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +{ + if (dai->dai_ops.digital_mute) + return dai->dai_ops.digital_mute(dai, mute); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- cgit v1.2.3