[alsa-devel] [PATCH 1/5] ASoC: twl4030 - AIF/APLL clock fix for all DAPM routes.
Peter Ujfalusi
peter.ujfalusi at nokia.com
Wed Apr 28 08:11:08 CEST 2010
Hi,
One comment
On Tuesday 27 April 2010 17:54:16 ext Liam Girdwood wrote:
> This allows the AIF and APLL clocks to continue running in
> loopback mode and when invalid DAPM routes are selected at
> stream runtime.
>
> Acked-by: Peter Ujfalusi <peter.ujfalusi at nokia.com>
> Acked-by: Mark Brown <broonie at opensource.wolfsonmicro.com>
> Signed-off-by: Liam Girdwood <lrg at slimlogic.co.uk>
> ---
> sound/soc/codecs/twl4030.c | 123
> +++++++++++++++++++++++++++++-------------- 1 files changed, 83
> insertions(+), 40 deletions(-)
>
> diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
> index dc26b71..1a3e7f6 100644
> --- a/sound/soc/codecs/twl4030.c
> +++ b/sound/soc/codecs/twl4030.c
> @@ -123,7 +123,10 @@ struct twl4030_priv {
> struct snd_soc_codec codec;
>
> unsigned int codec_powered;
> +
> + /* reference counts of AIF/APLL users */
> unsigned int apll_enabled;
> + unsigned int aif_enabled;
>
> struct snd_pcm_substream *master_substream;
> struct snd_pcm_substream *slave_substream;
> @@ -259,22 +262,50 @@ static void twl4030_init_chip(struct snd_soc_codec
> *codec) static void twl4030_apll_enable(struct snd_soc_codec *codec, int
> enable) {
> struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
> - int status;
> -
> - if (enable == twl4030->apll_enabled)
> - return;
> + int status = -1;
>
> + /* update the user count */
> if (enable)
> + twl4030->apll_enabled++;
> + else
> + twl4030->apll_enabled--;
> +
> + if (enable && twl4030->apll_enabled == 1)
> /* Enable PLL */
> status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL);
> - else
> + else if (!enable && twl4030->apll_enabled == 0)
> /* Disable PLL */
> status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL);
>
> if (status >= 0)
> twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
> +}
>
> - twl4030->apll_enabled = enable;
> +static void twl4030_aif_enable(struct snd_soc_codec *codec, int enable)
> +{
> + struct twl4030_priv *twl4030 = codec->private_data;
Should be to due the API change:
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
> +
> + /* update the user count */
> + if (enable)
> + twl4030->aif_enabled++;
> + else
> + twl4030->aif_enabled--;
> +
> + if (enable && twl4030->aif_enabled == 1) {
> + /* Enable AIF */
> + /* enable the PLL before we use it to clock the DAI */
> + twl4030_apll_enable(codec, 1);
> +
> + twl4030_write(codec, TWL4030_REG_AUDIO_IF,
> + twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF)
> + | TWL4030_AIF_EN);
> + } else if (!enable && twl4030->aif_enabled == 0) {
> + /* disable the DAI before we stop it's source PLL */
> + twl4030_write(codec, TWL4030_REG_AUDIO_IF,
> + twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF)
> + & ~TWL4030_AIF_EN);
> + twl4030_apll_enable(codec, 0);
> + }
> }
>
> static void twl4030_power_up(struct snd_soc_codec *codec)
> @@ -672,6 +703,20 @@ static int apll_event(struct snd_soc_dapm_widget *w,
> return 0;
> }
>
> +static int aif_event(struct snd_soc_dapm_widget *w,
> + struct snd_kcontrol *kcontrol, int event)
> +{
> + switch (event) {
> + case SND_SOC_DAPM_PRE_PMU:
> + twl4030_aif_enable(w->codec, 1);
> + break;
> + case SND_SOC_DAPM_POST_PMD:
> + twl4030_aif_enable(w->codec, 0);
> + break;
> + }
> + return 0;
> +}
> +
> static void headset_ramp(struct snd_soc_codec *codec, int ramp)
> {
> struct snd_soc_device *socdev = codec->socdev;
> @@ -1178,6 +1223,11 @@ static const struct snd_soc_dapm_widget
> twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"),
> SND_SOC_DAPM_OUTPUT("VIBRA"),
>
> + /* AIF and APLL clocks for running DAIs (including loopback) */
> + SND_SOC_DAPM_OUTPUT("AIF DAC"),
> + SND_SOC_DAPM_INPUT("AIF ADC"),
> + SND_SOC_DAPM_OUTPUT("APLL"),
> +
> /* DACs */
> SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
> SND_SOC_NOPM, 0, 0),
> @@ -1215,16 +1265,21 @@ static const struct snd_soc_dapm_widget
> twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassv_control),
>
> /* Digital mixers, power control for the physical DACs */
> - SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
> - TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0),
> - SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer",
> - TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0),
> - SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer",
> - TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0),
> - SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer",
> - TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0),
> - SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer",
> - TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0),
> + SND_SOC_DAPM_MIXER_E("Digital R1 Playback Mixer",
> + TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0, aif_event,
> + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_MIXER_E("Digital L1 Playback Mixer",
> + TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0, aif_event,
> + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_MIXER_E("Digital R2 Playback Mixer",
> + TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0, aif_event,
> + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_MIXER_E("Digital L2 Playback Mixer",
> + TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0, aif_event,
> + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
> + SND_SOC_DAPM_MIXER_E("Digital Voice Playback Mixer",
> + TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0, apll_event,
> + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
>
> /* Analog mixers, power control for the physical PGAs */
> SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer",
> @@ -1238,11 +1293,6 @@ static const struct snd_soc_dapm_widget
> twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog Voice Playback
> Mixer",
> TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
>
> - SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event,
> - SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
> -
> - SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0),
> -
> /* Output MIXER controls */
> /* Earpiece */
> SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
> @@ -1370,17 +1420,14 @@ static const struct snd_soc_dapm_route intercon[] =
> { {"Digital R2 Playback Mixer", NULL, "DAC Right2"},
> {"Digital Voice Playback Mixer", NULL, "DAC Voice"},
>
> - /* Supply for the digital part (APLL) */
> - {"Digital R1 Playback Mixer", NULL, "APLL Enable"},
> - {"Digital L1 Playback Mixer", NULL, "APLL Enable"},
> - {"Digital R2 Playback Mixer", NULL, "APLL Enable"},
> - {"Digital L2 Playback Mixer", NULL, "APLL Enable"},
> - {"Digital Voice Playback Mixer", NULL, "APLL Enable"},
> + /* Route for the digital part (APLL) */
> + {"APLL", NULL, "Digital Voice Playback Mixer"},
>
> - {"Digital R1 Playback Mixer", NULL, "AIF Enable"},
> - {"Digital L1 Playback Mixer", NULL, "AIF Enable"},
> - {"Digital R2 Playback Mixer", NULL, "AIF Enable"},
> - {"Digital L2 Playback Mixer", NULL, "AIF Enable"},
> + /* Route for digital part (AIF) */
> + {"AIF DAC", NULL, "Digital L1 Playback Mixer"},
> + {"AIF DAC", NULL, "Digital R1 Playback Mixer"},
> + {"AIF DAC", NULL, "Digital L2 Playback Mixer"},
> + {"AIF DAC", NULL, "Digital R2 Playback Mixer"},
>
> {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
> {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
> @@ -1493,15 +1540,11 @@ static const struct snd_soc_dapm_route intercon[] =
> { {"ADC Virtual Left2", NULL, "TX2 Capture Route"},
> {"ADC Virtual Right2", NULL, "TX2 Capture Route"},
>
> - {"ADC Virtual Left1", NULL, "APLL Enable"},
> - {"ADC Virtual Right1", NULL, "APLL Enable"},
> - {"ADC Virtual Left2", NULL, "APLL Enable"},
> - {"ADC Virtual Right2", NULL, "APLL Enable"},
> -
> - {"ADC Virtual Left1", NULL, "AIF Enable"},
> - {"ADC Virtual Right1", NULL, "AIF Enable"},
> - {"ADC Virtual Left2", NULL, "AIF Enable"},
> - {"ADC Virtual Right2", NULL, "AIF Enable"},
> + /* Route for digital part AIF */
> + {"ADC Virtual Left1", NULL, "AIF ADC"},
> + {"ADC Virtual Right1", NULL, "AIF ADC"},
> + {"ADC Virtual Left2", NULL, "AIF ADC"},
> + {"ADC Virtual Right2", NULL, "AIF ADC"},
>
> /* Analog bypass routes */
> {"Right1 Analog Loopback", "Switch", "Analog Right"},
--
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