[alsa-devel] [PATCH 1/7] ASoC: ad1836: 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 with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/ad1836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 685998d..95f0bec 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,7 +251,7 @@ static int ad1836_resume(struct snd_soc_codec *codec) static int ad1836_probe(struct snd_soc_codec *codec) { struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); int num_dacs, num_adcs; int ret = 0; int i;
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/adau1761.c | 26 ++++++++++++-------------- sound/soc/codecs/adau1781.c | 9 ++++----- sound/soc/codecs/adau17x1.c | 20 ++++++++++---------- 3 files changed, 26 insertions(+), 29 deletions(-)
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5ba2461..2f12477 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -482,6 +482,7 @@ static enum adau1761_output_mode adau1761_get_lineout_mode(
static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau1761_platform_data *pdata = codec->dev->platform_data; struct adau *adau = snd_soc_codec_get_drvdata(codec); enum adau1761_digmic_jackdet_pin_mode mode; @@ -514,21 +515,18 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec) if (ret) return ret; case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ - ret = snd_soc_dapm_add_routes(&codec->dapm, - adau1761_no_dmic_routes, + ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) return ret; break; case ADAU1761_DIGMIC_JACKDET_PIN_MODE_DIGMIC: - ret = snd_soc_dapm_new_controls(&codec->dapm, - adau1761_dmic_widgets, + ret = snd_soc_dapm_new_controls(dapm, adau1761_dmic_widgets, ARRAY_SIZE(adau1761_dmic_widgets)); if (ret) return ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm, - adau1761_dmic_routes, + ret = snd_soc_dapm_add_routes(dapm, adau1761_dmic_routes, ARRAY_SIZE(adau1761_dmic_routes)); if (ret) return ret; @@ -546,6 +544,7 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau *adau = snd_soc_codec_get_drvdata(codec); struct adau1761_platform_data *pdata = codec->dev->platform_data; enum adau1761_output_mode mode; @@ -576,12 +575,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec) }
if (mode == ADAU1761_OUTPUT_MODE_HEADPHONE_CAPLESS) { - ret = snd_soc_dapm_new_controls(&codec->dapm, + ret = snd_soc_dapm_new_controls(dapm, adau1761_capless_dapm_widgets, ARRAY_SIZE(adau1761_capless_dapm_widgets)); if (ret) return ret; - ret = snd_soc_dapm_add_routes(&codec->dapm, + ret = snd_soc_dapm_add_routes(dapm, adau1761_capless_dapm_routes, ARRAY_SIZE(adau1761_capless_dapm_routes)); } else { @@ -589,12 +588,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec) ARRAY_SIZE(adau1761_mono_controls)); if (ret) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, + ret = snd_soc_dapm_new_controls(dapm, adau1761_mono_dapm_widgets, ARRAY_SIZE(adau1761_mono_dapm_widgets)); if (ret) return ret; - ret = snd_soc_dapm_add_routes(&codec->dapm, + ret = snd_soc_dapm_add_routes(dapm, adau1761_mono_dapm_routes, ARRAY_SIZE(adau1761_mono_dapm_routes)); } @@ -639,6 +638,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg)
static int adau1761_codec_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau1761_platform_data *pdata = codec->dev->platform_data; struct adau *adau = snd_soc_codec_get_drvdata(codec); int ret; @@ -691,14 +691,12 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec) return ret;
if (adau->type == ADAU1761) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - adau1761_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, adau1761_dapm_widgets, ARRAY_SIZE(adau1761_dapm_widgets)); if (ret) return ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm, - adau1761_dapm_routes, + ret = snd_soc_dapm_add_routes(dapm, adau1761_dapm_routes, ARRAY_SIZE(adau1761_dapm_routes)); if (ret) return ret; diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 9c01ef0..fde9068 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -382,6 +382,7 @@ static int adau1781_set_input_mode(struct adau *adau, unsigned int reg,
static int adau1781_codec_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau1781_platform_data *pdata = dev_get_platdata(codec->dev); struct adau *adau = snd_soc_codec_get_drvdata(codec); int ret; @@ -402,19 +403,17 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) }
if (pdata && pdata->use_dmic) { - ret = snd_soc_dapm_new_controls(&codec->dapm, + ret = snd_soc_dapm_new_controls(dapm, adau1781_dmic_dapm_widgets, ARRAY_SIZE(adau1781_dmic_dapm_widgets)); if (ret) return ret; - ret = snd_soc_dapm_add_routes(&codec->dapm, - adau1781_dmic_dapm_routes, + ret = snd_soc_dapm_add_routes(dapm, adau1781_dmic_dapm_routes, ARRAY_SIZE(adau1781_dmic_dapm_routes)); if (ret) return ret; } else { - ret = snd_soc_dapm_add_routes(&codec->dapm, - adau1781_adc_dapm_routes, + ret = snd_soc_dapm_add_routes(dapm, adau1781_adc_dapm_routes, ARRAY_SIZE(adau1781_adc_dapm_routes)); if (ret) return ret; diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index fa2e690..fcf05b2 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -155,6 +155,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau *adau = snd_soc_codec_get_drvdata(codec); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct snd_soc_dapm_update update; @@ -188,7 +189,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, update.reg = reg; update.val = val;
- snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol, + snd_soc_dapm_mux_update_power(dapm, kcontrol, ucontrol->value.enumerated.item[0], e, &update); }
@@ -444,8 +445,8 @@ static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - struct snd_soc_dapm_context *dapm = &dai->codec->dapm;
switch (clk_id) { case ADAU17X1_CLK_SRC_MCLK: @@ -804,6 +805,7 @@ EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
int adau17x1_add_widgets(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau *adau = snd_soc_codec_get_drvdata(codec); int ret;
@@ -811,14 +813,13 @@ int adau17x1_add_widgets(struct snd_soc_codec *codec) ARRAY_SIZE(adau17x1_controls)); if (ret) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, adau17x1_dapm_widgets, ARRAY_SIZE(adau17x1_dapm_widgets)); if (ret) return ret;
if (adau17x1_has_dsp(adau)) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - adau17x1_dsp_dapm_widgets, + ret = snd_soc_dapm_new_controls(dapm, adau17x1_dsp_dapm_widgets, ARRAY_SIZE(adau17x1_dsp_dapm_widgets)); if (ret) return ret; @@ -840,21 +841,20 @@ EXPORT_SYMBOL_GPL(adau17x1_add_widgets);
int adau17x1_add_routes(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau *adau = snd_soc_codec_get_drvdata(codec); int ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes, + ret = snd_soc_dapm_add_routes(dapm, adau17x1_dapm_routes, ARRAY_SIZE(adau17x1_dapm_routes)); if (ret) return ret;
if (adau17x1_has_dsp(adau)) { - ret = snd_soc_dapm_add_routes(&codec->dapm, - adau17x1_dsp_dapm_routes, + ret = snd_soc_dapm_add_routes(dapm, adau17x1_dsp_dapm_routes, ARRAY_SIZE(adau17x1_dsp_dapm_routes)); } else { - ret = snd_soc_dapm_add_routes(&codec->dapm, - adau17x1_no_dsp_dapm_routes, + ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } return 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() and all remaining access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen lars@metafoo.de --- sound/soc/codecs/adau1977.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index e54d8d5..9bdd15f 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -485,7 +485,7 @@ static int adau1977_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 = adau1977_power_enable(adau1977); break; case SND_SOC_BIAS_OFF: @@ -848,12 +848,13 @@ static int adau1977_set_sysclk(struct snd_soc_codec *codec,
static int adau1977_codec_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); int ret;
switch (adau1977->type) { case ADAU1977: - ret = snd_soc_dapm_new_controls(&codec->dapm, + ret = snd_soc_dapm_new_controls(dapm, adau1977_micbias_dapm_widgets, ARRAY_SIZE(adau1977_micbias_dapm_widgets)); if (ret < 0)
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/adav80x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 260a652..36d8425 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -539,7 +539,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
if (dir == SND_SOC_CLOCK_IN) { switch (clk_id) { @@ -622,6 +622,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int pll_ctrl1 = 0; unsigned int pll_ctrl2 = 0; @@ -687,7 +688,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
adav80x->pll_src = source;
- snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync(dapm); }
return 0; @@ -800,11 +801,12 @@ static struct snd_soc_dai_driver adav80x_dais[] = {
static int adav80x_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
/* Force PLLs on for SYSCLK output */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + snd_soc_dapm_force_enable_pin(dapm, "PLL1"); + snd_soc_dapm_force_enable_pin(dapm, "PLL2");
/* Power down S/PDIF receiver, since it is currently not supported */ regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20);
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/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 13c6ab0..f30de76 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -510,7 +510,7 @@ static int ssm2518_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 = ssm2518_set_power(ssm2518, true); break; case SND_SOC_BIAS_OFF:
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/ssm2602.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 296a140..69a773a 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -523,8 +523,8 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
static int ssm2602_codec_probe(struct snd_soc_codec *codec) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret;
regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V, @@ -548,7 +548,7 @@ static int ssm2602_codec_probe(struct snd_soc_codec *codec)
static int ssm2604_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); int ret;
ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets,
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/ssm4567.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 643bcff..938d2cb 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -353,7 +353,7 @@ static int ssm4567_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 = ssm4567_set_power(ssm4567, true); break; case SND_SOC_BIAS_OFF:
On Mon, May 04, 2015 at 06:46:08PM +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 with snd_soc_codec_get_dapm().
Applied all, thanks.
participants (2)
-
Lars-Peter Clausen
-
Mark Brown