[alsa-devel] [PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- All patches in this series have a dependency on topic/dapm --- sound/soc/codecs/88pm860x-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index c0b2686..ee31fa7 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, break;
case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/ab8500-codec.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-)
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 88ca9cb..c7d243d 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1209,6 +1209,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; @@ -1234,15 +1235,14 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
- status = snd_soc_dapm_force_enable_pin(&codec->dapm, - "ANC Configure Input"); + status = snd_soc_dapm_force_enable_pin(dapm, "ANC Configure Input"); if (status < 0) { dev_err(dev, "%s: ERROR: Failed to enable power (status = %d)!\n", __func__, status); goto cleanup; } - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync(dapm);
anc_configure(codec, apply_fir, apply_iir);
@@ -1259,8 +1259,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, drvdata->anc_status = ANC_IIR_CONFIGURED; }
- status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); - snd_soc_dapm_sync(&codec->dapm); + status = snd_soc_dapm_disable_pin(dapm, "ANC Configure Input"); + snd_soc_dapm_sync(dapm);
cleanup: mutex_unlock(&drvdata->ctrl_lock); @@ -1947,6 +1947,7 @@ static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec) static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, struct amic_settings *amics) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); u8 value8; unsigned int value; int status; @@ -1973,15 +1974,15 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__, amic_micbias_str(amics->mic1a_micbias)); route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias]; - status = snd_soc_dapm_add_routes(&codec->dapm, route, 1); + status = snd_soc_dapm_add_routes(dapm, route, 1); dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__, amic_micbias_str(amics->mic1b_micbias)); route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias]; - status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + status |= snd_soc_dapm_add_routes(dapm, route, 1); dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__, amic_micbias_str(amics->mic2_micbias)); route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias]; - status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + status |= snd_soc_dapm_add_routes(dapm, route, 1); if (status < 0) { dev_err(codec->dev, "%s: Failed to add AMic-regulator DAPM-routes (%d).\n", @@ -2461,6 +2462,7 @@ static void ab8500_codec_of_probe(struct device *dev, struct device_node *np,
static int ab8500_codec_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct device *dev = codec->dev; struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); @@ -2541,7 +2543,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) &ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; drvdata->sid_fir_values = (long *)fc->value;
- (void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
mutex_init(&drvdata->ctrl_lock);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/ak4641.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 3b22b58..2d0ff45 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -412,7 +412,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20); break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { if (pdata && gpio_is_valid(pdata->gpio_power)) gpio_set_value(pdata->gpio_power, 1); mdelay(1);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/cx20442.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 13041cc..d6f4abb 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -333,7 +333,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
switch (level) { case SND_SOC_BIAS_PREPARE: - if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY) + if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_STANDBY) break; if (IS_ERR(cx20442->por)) err = PTR_ERR(cx20442->por); @@ -341,7 +341,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec, err = regulator_enable(cx20442->por); break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE) + if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_PREPARE) break; if (IS_ERR(cx20442->por)) err = PTR_ERR(cx20442->por);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/es8328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 996e3f4..6a09101 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -536,7 +536,7 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec, break;
case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { snd_soc_update_bits(codec, ES8328_CONTROL1, ES8328_CONTROL1_VMIDSEL_MASK | ES8328_CONTROL1_ENREF,
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/jz4740.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 8425d26..9363fdb 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -258,7 +258,7 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* The only way to clear the suspend flag is to reset the codec */ - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) jz4740_codec_wakeup(regmap);
mask = JZ4740_CODEC_1_VREF_DISABLE |
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/ml26124.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index f1d5778..62dda24 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -523,7 +523,7 @@ static int ml26124_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* VMID ON */ - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG, ML26124_VMID, ML26124_VMID); msleep(500);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/uda134x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index dbecbc0..913edf2 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -477,6 +477,7 @@ static struct snd_soc_dai_driver uda134x_dai = {
static int uda134x_soc_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->component.card->dev->platform_data; const struct snd_soc_dapm_widget *widgets; @@ -525,7 +526,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); }
- ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + ret = snd_soc_dapm_new_controls(dapm, widgets, num_widgets); if (ret) { printk(KERN_ERR "%s failed to register dapm controls: %d", __func__, ret);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Also drop the unnecessary check at the beginning of the uda1380_set_bias_level() which compares the current level to the target level and aborts if they are the same. Since the core will not call the set_bias_level() callback if we already are in the target state the result of the check is always false.
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/uda1380.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f761a2..6e159f5 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -590,9 +590,6 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, int reg; struct uda1380_platform_data *pdata = codec->dev->platform_data;
- if (codec->dapm.bias_level == level) - return 0; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -600,7 +597,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm); break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { if (gpio_is_valid(pdata->gpio_power)) { gpio_set_value(pdata->gpio_power, 1); mdelay(1);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 69bc87d..e673f6c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -948,7 +948,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { ret = regulator_bulk_enable( ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies);
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/sirf-audio-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 0a8e43c..29cb442 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -395,7 +395,7 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = {
static int sirf_audio_codec_probe(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
pm_runtime_enable(codec->dev);
On Mon, May 11, 2015 at 09:42:26AM +0200, Lars-Peter Clausen wrote:
The dapm field of the snd_soc_codec struct is eventually going to be removed, in preparation for this replace all manual access to codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Applied all, thanks.
participants (2)
-
Lars-Peter Clausen -
Mark Brown