October 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/11 v2] Convert wm8xxx codec drivers to use snd_soc_write/snd_soc_cache_sync
by Axel Lin 07 Oct '11

07 Oct '11

This serial converts the wm8xxx codec drivers to use snd_soc_cache_sync. I don't have these hardware, but I assume Mark has all these hardware. These patches are UNTESTED. I'd appreciate if someone can test it. (Current code is broken anyway, thus we need convert to use new API. ) v2: For 7, 9 CODECs that are write only, convert to using cache_bypass + snd_soc_write v2 also includes wm8990 that I had sent separately. *** BLURB HERE *** Axel Lin (11): ASoC: wm8510: Convert to snd_soc_write ASoC: wm8711: Convert to snd_soc_write ASoC: wm8731: Convert to snd_soc_write ASoC: wm8750: Convert to snd_soc_write ASoC: wm8776: Convert to snd_soc_write ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_write ASoC: wm8971: Convert to snd_soc_write ASoC: wm8974: Convert to snd_soc_write ASoC: wm8988: Convert to snd_soc_write ASoC: wm8990: Convert to snd_soc_cache_sync sound/soc/codecs/wm8510.c | 33 +++++++++++++++++++++---------- sound/soc/codecs/wm8711.c | 33 +++++++++++++++++++++---------- sound/soc/codecs/wm8731.c | 35 +++++++++++++++++++++------------ sound/soc/codecs/wm8750.c | 36 +++++++++++++++++++++------------- sound/soc/codecs/wm8776.c | 36 +++++++++++++++++++++------------- sound/soc/codecs/wm8940.c | 46 +++++++++++++++++++++----------------------- sound/soc/codecs/wm8960.c | 34 +++++++++++++++++++++++--------- sound/soc/codecs/wm8971.c | 35 ++++++++++++++++++++++----------- sound/soc/codecs/wm8974.c | 33 +++++++++++++++++++++---------- sound/soc/codecs/wm8988.c | 37 +++++++++++++++++++++-------------- sound/soc/codecs/wm8990.c | 32 ++++++++++++++++++------------ 11 files changed, 242 insertions(+), 148 deletions(-) -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: wm_hubs: Remove unneeded count variable in wait_for_dc_servo
by Axel Lin 07 Oct '11

07 Oct '11

The count variable is not required, decrement timeout in the while loop can achieve the same effect. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm_hubs.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4..6e2e122 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -66,7 +66,6 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; - int count = 0; int timeout; unsigned int val; @@ -83,7 +82,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) timeout = 400; do { - count++; + timeout--; if (hubs->dcs_done_irq) wait_for_completion_timeout(&hubs->dcs_done, @@ -93,9 +92,9 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) reg = snd_soc_read(codec, WM8993_DC_SERVO_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < timeout); + } while ((reg & op) && timeout); - if (reg & op) + if (timeout == 0) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", op); } -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: am3517evm: Let core to deal with the DAPM widgets
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Pass the DAPM widgets/routes via the snd_soc_card struct to core. With this change we do not need the init function since the remaining snd_soc_dapm_enable_pin calls are not needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/am3517evm.c | 28 +++++----------------------- 1 files changed, 5 insertions(+), 23 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index f9b995f..8da55e9 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -90,28 +90,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic In"}, }; -static int am3517evm_aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add am3517-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic In"); - - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link am3517evm_dai = { .name = "TLV320AIC23", @@ -122,7 +100,6 @@ static struct snd_soc_dai_link am3517evm_dai = { .codec_name = "tlv320aic23-codec.2-001a", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; @@ -131,6 +108,11 @@ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", .dai_link = &am3517evm_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *am3517evm_snd_device; -- 1.7.7

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[alsa-devel] [PATCH] ASoC: osk5912: Let core to deal with the DAPM widgets
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Pass the DAPM widgets/routes via the snd_soc_card struct to core. With this change we do not need the init function since the remaining snd_soc_dapm_enable_pin calls are not needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/osk5912.c | 27 +++++---------------------- 1 files changed, 5 insertions(+), 22 deletions(-) diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 4a13c45..db91cca 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -91,27 +91,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int osk_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add osk5912 specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up osk5912 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", @@ -122,7 +101,6 @@ static struct snd_soc_dai_link osk_dai = { .codec_name = "tlv320aic23-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, - .init = osk_tlv320aic23_init, .ops = &osk_ops, }; @@ -131,6 +109,11 @@ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", .dai_link = &osk_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *osk_snd_device; -- 1.7.7

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[alsa-devel] [PATCH] ASoC: n810: Let the core to register DAPM widgets/routes and controls
by Peter Ujfalusi 07 Oct '11

07 Oct '11

Pass the DAPM widgets/routes and static controls via the snd_soc_card struct to core. In this way the machine driver does not need to handle the DAPM widgets/routes. Mark the not connected pins at late_probe time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/n810.c | 47 ++++++++++++++++++----------------------------- 1 files changed, 18 insertions(+), 29 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index bda2aa4..625e886 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -253,36 +253,16 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { n810_get_input, n810_set_input), }; -static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) +static int n810_aic33_late_probe(struct snd_soc_card *card) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; - - /* Not connected */ - snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(dapm, "HPLCOM"); - snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - snd_soc_dapm_nc_pin(dapm, "MIC3L"); - snd_soc_dapm_nc_pin(dapm, "MIC3R"); - snd_soc_dapm_nc_pin(dapm, "LINE1R"); - snd_soc_dapm_nc_pin(dapm, "LINE2L"); - snd_soc_dapm_nc_pin(dapm, "LINE2R"); - - /* Add N810 specific controls */ - err = snd_soc_add_controls(codec, aic33_n810_controls, - ARRAY_SIZE(aic33_n810_controls)); - if (err < 0) - return err; - - /* Add N810 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic33_dapm_widgets, - ARRAY_SIZE(aic33_dapm_widgets)); - - /* Set up N810 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_dapm_new_widgets(dapm); + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "MIC3L"); + snd_soc_dapm_nc_pin(&card->dapm, "MIC3R"); + snd_soc_dapm_nc_pin(&card->dapm, "LINE1R"); + snd_soc_dapm_nc_pin(&card->dapm, "LINE2L"); + snd_soc_dapm_nc_pin(&card->dapm, "LINE2R"); return 0; } @@ -306,6 +286,15 @@ static struct snd_soc_card snd_soc_n810 = { .name = "N810", .dai_link = &n810_dai, .num_links = 1, + + .controls = aic33_n810_controls, + .num_controls = ARRAY_SIZE(aic33_n810_controls), + .dapm_widgets = aic33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), + + .late_probe = n810_aic33_late_probe, }; static struct platform_device *n810_snd_device; -- 1.7.7

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[alsa-devel] [PATCH 1/2] ASoC: sdp440: No need to call snd_soc_dapm_enable_pin at init time
by Peter Ujfalusi 07 Oct '11

07 Oct '11

All DAPM widgets are connected after they have added. Machine driver only need to mark not connected pins. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/sdp4430.c | 8 -------- 1 files changed, 0 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 10972fb..3069226 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -132,14 +132,6 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Set up SDP4430 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* SDP4430 connected pins */ - snd_soc_dapm_enable_pin(dapm, "Ext Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - snd_soc_dapm_enable_pin(dapm, "AFML"); - snd_soc_dapm_enable_pin(dapm, "AFMR"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - ret = snd_soc_dapm_new_widgets(dapm); if (ret) return ret; -- 1.7.7

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[alsa-devel] [PATCH] soc: codecs: tlv320aic32x4: fix initialization of micpga routing
by Wolfram Sang 06 Oct '11

06 Oct '11

Checking the pdata-flags used 'or', so the check is always true. Use 'and' to correctly mask the flags. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Javier Martin <javier.martin(a)vista-silicon.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e93b9d1..a68982e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -685,10 +685,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } /* Mic PGA routing */ - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); } - if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); } -- 1.7.2.5

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[alsa-devel] [PATCH 00/10] Convert wm8xxx codec drivers to use snd_soc_cache_sync
by Axel Lin 06 Oct '11

06 Oct '11

This serial converts the wm8xxx codec drivers to use snd_soc_cache_sync. I don't have these hardware, but I assume Mark has all these hardware. These patches are UNTESTED. I'd appreciate if someone can test them. (Current code is broken anyway, thus we need convert to use new API. ) Axel Lin (10): ASoC: wm8510: Convert to snd_soc_cache_sync ASoC: wm8711: Convert to snd_soc_cache_sync ASoC: wm8731: Convert to snd_soc_cache_sync ASoC: wm8750: Convert to snd_soc_cache_sync ASoC: wm8776: Convert to snd_soc_cache_sync ASoC: wm8940: Convert to snd_soc_cache_sync ASoC: wm8960: Convert to snd_soc_cache_sync ASoC: wm8971: Convert to snd_soc_cache_sync ASoC: wm8974: Convert to snd_soc_cache_sync ASoC: wm8988: Convert to snd_soc_cache_sync sound/soc/codecs/wm8510.c | 30 ++++++++++++++++++---------- sound/soc/codecs/wm8711.c | 19 +++++++---------- sound/soc/codecs/wm8731.c | 29 ++++++++++++++++----------- sound/soc/codecs/wm8750.c | 32 +++++++++++++++++------------- sound/soc/codecs/wm8776.c | 33 ++++++++++++++++++------------- sound/soc/codecs/wm8940.c | 46 +++++++++++++++++++++----------------------- sound/soc/codecs/wm8960.c | 37 +++++++++++++++++++++++++---------- sound/soc/codecs/wm8971.c | 32 +++++++++++++++++++----------- sound/soc/codecs/wm8974.c | 29 +++++++++++++++++---------- sound/soc/codecs/wm8988.c | 32 +++++++++++++++++------------- 10 files changed, 185 insertions(+), 134 deletions(-) -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Ensure DAPM widgets are set up before we sync jacks
by Mark Brown 06 Oct '11

06 Oct '11

We synchronise jack state on startup - when we do that make sure that we have set up all the DAPM widgets first in case we end up touching any of the partially set up widgets when syncing the jack pins. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 02f95aa..6c5ebd3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -189,6 +189,8 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, list_add(&(pins[i].list), &jack->pins); } + snd_soc_dapm_new_widgets(&jack->codec->card->dapm); + /* Update to reflect the last reported status; canned jack * implementations are likely to set their state before the * card has an opportunity to associate pins. -- 1.7.6.3

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[alsa-devel] [PATCH v2] ASoC: Return early with -EINVAL if invalid dai format is detected
by Axel Lin 06 Oct '11

06 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Timur Tabi <timur(a)freescale.com> Cc: Arnaud Patard <arnaud.patard(a)rtp-net.org> --- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs42l51.c | 9 ++++----- 2 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5830c93..f1f237e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -261,7 +261,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int ret = 0; /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -271,7 +270,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid dai format\n"); - ret = -EINVAL; + return -EINVAL; } /* set master/slave audio interface */ @@ -284,10 +283,11 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: /* all other modes are unsupported by the hardware */ - ret = -EINVAL; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } /** diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 286878d..8c3c820 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -288,7 +288,6 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); - int ret = 0; switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -298,7 +297,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, break; default: dev_err(codec->dev, "invalid DAI format\n"); - ret = -EINVAL; + return -EINVAL; } switch (format & SND_SOC_DAIFMT_MASTER_MASK) { @@ -309,11 +308,11 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai, cs42l51->func = MODE_SLAVE_AUTO; break; default: - ret = -EINVAL; - break; + dev_err(codec->dev, "Unknown master/slave configuration\n"); + return -EINVAL; } - return ret; + return 0; } struct cs42l51_ratios { -- 1.7.4.1

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