[alsa-devel] [PATCH 1/3] ASoC: Implement WM8903 oversampling rate controls
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/wm8903.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 797992c..542c7c4 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -481,6 +481,16 @@ static const char *hpf_mode_text[] = { static const struct soc_enum hpf_mode = SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
+static const char *osr_text[] = { + "Low power", "High performance" +}; + +static const struct soc_enum adc_osr = + SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text); + +static const struct soc_enum dac_osr = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text); + static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; @@ -619,6 +629,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, 6, 1, 0),
/* ADCs */ +SOC_ENUM("ADC OSR", adc_osr), SOC_SINGLE("HPF Switch", WM8903_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("HPF Mode", hpf_mode), SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), @@ -650,6 +661,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8, 12, 0, digital_sidetone_tlv),
/* DAC */ +SOC_ENUM("DAC OSR", dac_osr), SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT, WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv), SOC_ENUM("DAC Soft Mute Rate", soft_mute),
We're already flagged as using symmetric rates so we don't need to have a custom implementation.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/wm8903.c | 79 +-------------------------------------------- 1 files changed, 1 insertions(+), 78 deletions(-)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 542c7c4..f9ae403 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -218,10 +218,8 @@ struct wm8903_priv { int sysclk; int irq;
- /* Reference counts */ + /* Reference count */ int class_w_users; - int playback_active; - int capture_active;
struct completion wseq;
@@ -230,9 +228,6 @@ struct wm8903_priv { int mic_short; int mic_last_report; int mic_delay; - - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; };
static int wm8903_volatile_register(unsigned int reg) @@ -1243,58 +1238,6 @@ static struct { { 0, 0 }, };
-static int wm8903_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - wm8903->playback_active++; - else - wm8903->capture_active++; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - */ - if (wm8903->master_substream) { - master_runtime = wm8903->master_substream->runtime; - - dev_dbg(codec->dev, "Constraining to %d bits\n", - master_runtime->sample_bits); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - wm8903->slave_substream = substream; - } else - wm8903->master_substream = substream; - - return 0; -} - -static void wm8903_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - wm8903->playback_active--; - else - wm8903->capture_active--; - - if (wm8903->master_substream == substream) - wm8903->master_substream = wm8903->slave_substream; - - wm8903->slave_substream = NULL; -} - static int wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1319,11 +1262,6 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, u16 clock1 = snd_soc_read(codec, WM8903_CLOCK_RATES_1); u16 dac_digital1 = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
- if (substream == wm8903->slave_substream) { - dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - /* Enable sloping stopband filter for low sample rates */ if (fs <= 24000) dac_digital1 |= WM8903_DAC_SB_FILT; @@ -1341,19 +1279,6 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, } }
- /* Constraints should stop us hitting this but let's make sure */ - if (wm8903->capture_active) - switch (sample_rates[dsp_config].rate) { - case 88200: - case 96000: - dev_err(codec->dev, "%dHz unsupported by ADC\n", - fs); - return -EINVAL; - - default: - break; - } - dev_dbg(codec->dev, "DSP fs = %dHz\n", sample_rates[dsp_config].rate); clock1 &= ~WM8903_SAMPLE_RATE_MASK; clock1 |= sample_rates[dsp_config].value; @@ -1592,8 +1517,6 @@ static irqreturn_t wm8903_irq(int irq, void *data) SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops wm8903_dai_ops = { - .startup = wm8903_startup, - .shutdown = wm8903_shutdown, .hw_params = wm8903_hw_params, .digital_mute = wm8903_digital_mute, .set_fmt = wm8903_set_dai_fmt,
On Fri, 2010-12-10 at 19:17 +0000, Mark Brown wrote:
We're already flagged as using symmetric rates so we don't need to have a custom implementation.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
sound/soc/codecs/wm8903.c | 79 +-------------------------------------------- 1 files changed, 1 insertions(+), 78 deletions(-)
Acked-by: Liam Girdwood lrg@slimlogic.co.uk
Provide the user with a boolean control then automatically select the deemphasis filter most closely matching the sample rate.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/wm8903.c | 82 ++++++++++++++++++++++++++++++++++++++++---- 1 files changed, 74 insertions(+), 8 deletions(-)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index f9ae403..d015745 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -218,6 +218,9 @@ struct wm8903_priv { int sysclk; int irq;
+ int fs; + int deemph; + /* Reference count */ int class_w_users;
@@ -457,6 +460,72 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+static int wm8903_deemph[] = { 0, 32000, 44100, 48000 }; + +static int wm8903_set_deemph(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int val, i, best; + + /* If we're using deemphasis select the nearest available sample + * rate. + */ + if (wm8903->deemph) { + best = 1; + for (i = 2; i < ARRAY_SIZE(wm8903_deemph); i++) { + if (abs(wm8903_deemph[i] - wm8903->fs) < + abs(wm8903_deemph[best] - wm8903->fs)) + best = i; + } + + val = best << WM8903_DEEMPH_SHIFT; + } else { + best = 0; + val = 0; + } + + dev_dbg(codec->dev, "Set deemphasis %d (%dHz)\n", + best, wm8903_deemph[best]); + + return snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1, + WM8903_DEEMPH_MASK, val); +} + +static int wm8903_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8903->deemph; + + return 0; +} + +static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + int ret = 0; + + if (deemph > 1) + return -EINVAL; + + mutex_lock(&codec->mutex); + if (wm8903->deemph != deemph) { + wm8903->deemph = deemph; + + wm8903_set_deemph(codec); + + ret = 1; + } + mutex_unlock(&codec->mutex); + + return ret; +} + /* ALSA can only do steps of .01dB */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
@@ -548,13 +617,6 @@ static const char *mute_mode_text[] = { static const struct soc_enum mute_mode = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text);
-static const char *dac_deemphasis_text[] = { - "Disabled", "32kHz", "44.1kHz", "48kHz" -}; - -static const struct soc_enum dac_deemphasis = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 1, 4, dac_deemphasis_text); - static const char *companding_text[] = { "ulaw", "alaw" }; @@ -662,9 +724,10 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT, SOC_ENUM("DAC Soft Mute Rate", soft_mute), SOC_ENUM("DAC Mute Mode", mute_mode), SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0), -SOC_ENUM("DAC De-emphasis", dac_deemphasis), SOC_ENUM("DAC Companding Mode", dac_companding), SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0), +SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0, + wm8903_get_deemph, wm8903_put_deemph),
/* Headphones */ SOC_DOUBLE_R("Headphone Switch", @@ -1374,6 +1437,9 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, aif2 |= bclk_divs[bclk_div].div; aif3 |= bclk / fs;
+ wm8903->fs = params_rate(params); + wm8903_set_deemph(codec); + snd_soc_write(codec, WM8903_CLOCK_RATES_0, clock0); snd_soc_write(codec, WM8903_CLOCK_RATES_1, clock1); snd_soc_write(codec, WM8903_AUDIO_INTERFACE_1, aif1);
On Fri, 2010-12-10 at 19:17 +0000, Mark Brown wrote:
Provide the user with a boolean control then automatically select the deemphasis filter most closely matching the sample rate.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
sound/soc/codecs/wm8903.c | 82 ++++++++++++++++++++++++++++++++++++++++---- 1 files changed, 74 insertions(+), 8 deletions(-)
Acked-by: Liam Girdwood lrg@slimlogic.co.uk
On Fri, 2010-12-10 at 19:17 +0000, Mark Brown wrote:
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
sound/soc/codecs/wm8903.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-)
Acked-by: Liam Girdwood lrg@slimlogic.co.uk
participants (2)
-
Liam Girdwood -
Mark Brown