January 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 27 Jan '12

27 Jan '12

On Fri, Jan 27, 2012 at 05:46:02AM -0800, Reddy, MR Swami wrote: Overall this is a substantial improvement on your last version but there's still quite a few issues here. Looking at some of the issues I'm concerned about how you're testing this, there are some things here that I'd expect the core to complain about when you load the driver. > +/* Register default values for LM49453 river. */ > +static const u8 lm49453_reg_defs[LM49453_MAX_REGISTER] = { Please use the regmap API as previously requested. If anything this is going backwards as you've used a big table rather than [reg] = value but really, use the regmap API. The long term goal is to remove the register cache from ASoC completely. > +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { > +SOC_DAPM_SINGLE("Port1_1 HPL Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), These controls are all going to end up called "HPL Mixer Port1_1 HPL Switch" and so on which probably isn't what you want. You shouldn't have the "HPL" in any of the individual mixer controls, the prefixing is enough to let the user know this is affecting the HPL mixer. The same issue applies to all your mixer controls. > +static const struct snd_kcontrol_new lm49453_snd_controls[] = { > + SOC_DAPM_ENUM("Mic2Mode Capture Route", lm49453_mic2mode_enum), > + SOC_DAPM_ENUM("DMIC12 Capture Route", lm49453_dmic12_cfg_enum), > + SOC_DAPM_ENUM("DMIC34 Capture Route", lm49453_dmic34_cfg_enum), Why are these appearing as regular controls? DAPM controls should be attached to widgets... > + /* Sidetone supports mono only */ > + SOC_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, > + 0, 0x3F, 0, digital_tlv), These should probably line up with the relevant mixer controls. > + SND_SOC_DAPM_INPUT("AMIC1"), /* Headset mic */ Why is this specifically the headset mic? > + /* HP mapping */ > + { "HPL Mixer", "PORT1_1_RX_LVL Volume", "PORT1_1_RX" }, > + { "HPL Mixer", "PORT1_2_RX_LVL Volume", "PORT1_2_RX" }, > + { "HPL Mixer", "PORT1_3_RX_LVL Volume", "PORT1_3_RX"}, > + { "HPL Mixer", "PORT1_4_RX_LVL Volume", "PORT1_4_RX" }, > + { "HPL Mixer", "PORT1_5_RX_LVL Volume", "PORT1_5_RX" }, > + { "HPL Mixer", "PORT1_6_RX_LVL Volume", "PORT1_6_RX" }, > + { "HPL Mixer", "PORT1_7_RX_LVL Volume", "PORT1_7_RX" }, > + { "HPL Mixer", "PORT1_8_RX_LVL Volume", "PORT1_8_RX" }, This doesn't line up with the switches that are in the DAPM controls for the mixers, the Volume controls are nothing to do with DAPM. > + { "Port2_1 Mixer", "Port1_1 Port2_1 Switch", "PORT1_1_RX_LVL Volume" }, PORT1_1_RX_LVL Volume is not a DAPM widget. I'm very surprised this driver manages to register without generating errors, how have you tested this? > + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQL_REG, > + LM49453_FLL_REF_FREQ_VAL & 0xff); > + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQH_REG, > + (LM49453_FLL_REF_FREQ_VAL >> 8) & 0xff); > + > + snd_soc_write(codec, LM49453_P0_VCO_TARGETLL_REG, > + LM49453_VCO_TARGET_VAL & 0xff); > + snd_soc_write(codec, LM49453_P0_VCO_TARGETLH_REG, > + (LM49453_VCO_TARGET_VAL >> 8) & 0xff); > + snd_soc_write(codec, LM49453_P0_VCO_TARGETHL_REG, > + (LM49453_VCO_TARGET_VAL >> 16) & 0xff); > + snd_soc_write(codec, LM49453_P0_VCO_TARGETHH_REG, > + (LM49453_VCO_TARGET_VAL >> 24) & 0xff); This appears to completely ignore the parameters the user passed in, how can this ever work? > +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, > + unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); > + > + switch (freq) { > + case 12288000: /* Platform System Clock */ What does this comment mean? > +static int lm49453_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *codec_dai) > +{ > + unsigned int startup_mask; > + struct snd_soc_codec *codec = codec_dai->codec; > + dev_dbg(codec->dev, "lm49453_startup\n"); > + > + startup_mask = LM49453_PMC_SETUP_CHIP_EN | LM49453_PMC_SETUP_PLL_EN; > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, startup_mask, > + startup_mask); This looks like it should be DAPM or bias level manaement, and indeed CHIP_EN is disabled by bias management. > +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) > +{ > + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), > + BIT(1)|BIT(0)); > + return 0; > +} This completely ignores the mute parameter. > +static int __init lm49453_modinit(void) > +{ > + int ret; > + ret = i2c_add_driver(&lm49453_i2c_driver); > + if (ret != 0) > + pr_err("Failed to register LM49453 I2C driver: %d\n", ret); > + > + return ret; > +} > +module_init(lm49453_modinit); Use module_i2c_driver.

1 0

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Asus N53Jq
by David Henningsson 27 Jan '12

27 Jan '12

The user reports that he needs to add model=auto for audio to work properly. In fact, since node 0x15 is not even a pin node, the existing fixup is definitely wrong. Relevant information can be found in the buglink below. Cc: stable(a)kernel.org (3.2+) BugLink: https://bugs.launchpad.net/bugs/918254 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0db1dc4..a7f17be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5377,7 +5377,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC), -- 1.7.8.3

2 1

[alsa-devel] ASoC davinci: playing mono sound doubles all samples on the i2s (pcm) bus except the first one
by Jan Pohanka 27 Jan '12

27 Jan '12

Hello, I'm working with a board with TI dm365 processor and using McBSP pheripheral to send sound samples to a GSM module. The kernel version is 2.6.32.17 and the ASoC driver state is based on this commit http://arago-project.org/git/projects/?p=linux-davinci.git;a=commit;h=a4c8e… I have configured the hw interface to send two words in a frame on 8000Hz, this is working fine. When playing mono waveforms I noticed that all samples are doubled (it should be correct because snd_pcm_hardware has SNDRV_PCM_INFO_INTERLEAVED flag) but except the first one, that is send only once. This leads to a state when one sample in each frame is delayed. Unfortunately I'm not an expert on ALSA and ASoC and I'm not sure whether this problem rises in the core of ALSA or only in the driver for davinci dm365 (probably somewhere in the dma handling). Could please someone give me some hint where to find the source of the problem? I hope I described the situation enough... best regards Jan -- Tato zpráva byla vytvořena převratným poštovním klientem Opery: http://www.opera.com/mail/

1 0

[alsa-devel] [PATCH] ASoC: Set idle_bias_off flag in snd_soc_codec_driver
by Axel Lin 27 Jan '12

27 Jan '12

Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver for devices can unconditionally support idle_bias_off. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Harsha Priya <priya.harsha(a)intel.com> Cc: Vladimir Barinov <vova.barinov(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Steve Sakoman <steve(a)sakoman.com> Cc: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/adau1373.c | 3 +-- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/sn95031.c | 3 +-- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/wm8770.c | 3 +-- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 4 +--- 11 files changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 971ba45..facda33 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1244,8 +1244,6 @@ static int adau1373_probe(struct snd_soc_codec *codec) return ret; } - codec->dapm.idle_bias_off = true; - if (pdata) { if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) return -EINVAL; @@ -1340,6 +1338,7 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), .reg_cache_default = adau1373_default_regs, .reg_word_size = sizeof(uint8_t), diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 6b325ea..78e9ce4 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -457,7 +457,6 @@ static int adau1701_probe(struct snd_soc_codec *codec) { int ret; - codec->dapm.idle_bias_off = 1; codec->control_data = to_i2c_client(codec->dev); ret = adau1701_load_firmware(codec); @@ -473,6 +472,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver adau1701_codec_drv = { .probe = adau1701_probe, .set_bias_level = adau1701_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ADAU1701_NUM_REGS, .reg_word_size = sizeof(u16), diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index f99baa0..aa03923 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -827,8 +827,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - codec->dapm.idle_bias_off = 1; - /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -891,6 +889,7 @@ struct snd_soc_codec_driver sn95031_codec = { .read = sn95031_read, .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, + .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 492f22f..285b7a2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1377,7 +1377,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); if (ret != 0) { @@ -1471,6 +1470,7 @@ static int aic3x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(aic3x_reg), .reg_word_size = sizeof(u8), .reg_cache_default = aic3x_reg, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c06c3e4..2c957c8 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1395,7 +1395,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->dapm.idle_bias_off = 1; dac33->codec = codec; /* Read the tlv320dac33 ID registers */ @@ -1476,6 +1475,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, .set_bias_level = dac33_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(dac33_reg), .reg_word_size = sizeof(u8), .reg_cache_default = dac33_reg, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a193f5f..3039ba2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2227,7 +2227,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_audio_get_mclk() / 1000; - codec->dapm.idle_bias_off = 1; twl4030_init_chip(codec); @@ -2253,6 +2252,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, + .idle_bias_off = true, .reg_cache_size = sizeof(twl4030_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl4030_reg, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19374a9..bd60f84 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,8 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - codec->dapm.idle_bias_off = 1; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -679,6 +677,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .suspend = wm8770_suspend, .resume = wm8770_resume, .set_bias_level = wm8770_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(wm8770_reg_defs), .reg_word_size = sizeof (u16), .reg_cache_default = wm8770_reg_defs diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 8abe375..7ee8dcf 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -549,7 +549,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); - codec->dapm.idle_bias_off = 1; codec->control_data = wm8804->regmap; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); @@ -678,6 +677,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .suspend = wm8804_suspend, .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, + .idle_bias_off = true, .controls = wm8804_snd_controls, .num_controls = ARRAY_SIZE(wm8804_snd_controls), diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 14afc11..37079ea 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2088,7 +2088,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) int ret, i; codec->cache_sync = 1; - codec->dapm.idle_bias_off = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2237,6 +2236,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .suspend = wm8904_suspend, .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, + .idle_bias_off = true, }; static const struct regmap_config wm8904_regmap = { diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index c8aada5..89a8642 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2047,7 +2047,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) int i; int ret; - codec->dapm.idle_bias_off = 1; wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; @@ -2241,6 +2240,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .suspend = wm8995_suspend, .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, + .idle_bias_off = true, }; static struct regmap_config wm8995_regmap = { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c70b7e6..f41e17a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2805,7 +2805,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) int ret; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - struct snd_soc_dapm_context *dapm = &codec->dapm; int i, irq_flags; wm8996->codec = codec; @@ -2813,8 +2812,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->dcs_done); init_completion(&wm8996->fll_lock); - dapm->idle_bias_off = true; - codec->control_data = wm8996->regmap; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); @@ -3068,6 +3065,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .probe = wm8996_probe, .remove = wm8996_remove, .set_bias_level = wm8996_set_bias_level, + .idle_bias_off = true, .seq_notifier = wm8996_seq_notifier, .controls = wm8996_snd_controls, .num_controls = ARRAY_SIZE(wm8996_snd_controls), -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ALSA: Convert at73c213 to module_spi_driver
by Axel Lin 27 Jan '12

27 Jan '12

This patch converts at73c213 to use the module_spi_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/spi/at73c213.c | 12 +----------- 1 files changed, 1 insertions(+), 11 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd051b..c6500d0 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1112,17 +1112,7 @@ static struct spi_driver at73c213_driver = { .remove = __devexit_p(snd_at73c213_remove), }; -static int __init at73c213_init(void) -{ - return spi_register_driver(&at73c213_driver); -} -module_init(at73c213_init); - -static void __exit at73c213_exit(void) -{ - spi_unregister_driver(&at73c213_driver); -} -module_exit(at73c213_exit); +module_spi_driver(at73c213_driver); MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt(a)samfundet.no>"); MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); -- 1.7.5.4

1 0

[alsa-devel] [PATCH] ALSA: aoa: Convert onyx and tas codec drivers to module_i2c_driver
by Axel Lin 27 Jan '12

27 Jan '12

This patch converts onyx and tas codec drivers to use the module_i2c_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/aoa/codecs/onyx.c | 13 +------------ sound/aoa/codecs/tas.c | 13 +------------ 2 files changed, 2 insertions(+), 24 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 762af68..270790d 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1132,15 +1132,4 @@ static struct i2c_driver onyx_driver = { .id_table = onyx_i2c_id, }; -static int __init onyx_init(void) -{ - return i2c_add_driver(&onyx_driver); -} - -static void __exit onyx_exit(void) -{ - i2c_del_driver(&onyx_driver); -} - -module_init(onyx_init); -module_exit(onyx_exit); +module_i2c_driver(onyx_driver); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index fd2188c..8e63d1f 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -1026,15 +1026,4 @@ static struct i2c_driver tas_driver = { .id_table = tas_i2c_id, }; -static int __init tas_init(void) -{ - return i2c_add_driver(&tas_driver); -} - -static void __exit tas_exit(void) -{ - i2c_del_driver(&tas_driver); -} - -module_init(tas_init); -module_exit(tas_exit); +module_i2c_driver(tas_driver); -- 1.7.5.4

1 0

[alsa-devel] Problems with rate plugin and writing sizes other than the period size
by Trent Piepho 27 Jan '12

27 Jan '12

This is a problem I've been able to duplicate on multiple systems with different hardware. If one sets up rate conversion, from 22050 Hz to 48000 Hz, which is a non-integer ratio, then a hw params with two periods doesn't work if you write greater than the period size. An audio underrun occurs after each period. Tracing with both strace and inside the driver reveals what's going on. The first snd_pcm_writei() goes through and the driver receives the appropriate number of rate converted frames. On the next snd_pcm_writei() alsa-lib calls poll() to wait for space to be available in the buffer. avail_min is set to one period, and there is less than one period left in the buffer, since the first write wrote more than one period. Eventually avail is greater than avail_min and poll returns. So far as expected. Now here's the problem, rather than writing a period alsa-lib just calls poll() again. Which returns immediately since there is still greater than avail_min available space in the buffer. And this continues, with alsa-lib in a tight loop calling poll() and poll() returning immediately, until the buffer empties and underflow occurs. If the number of periods is increased the same thing happens, without the underflow each period. Alsa-lib will wait until there are TWO empty periods in the buffer, in a tight loop calling poll() which returns immediately while avail is greater avail_min and less than two periods worth. It's basically like the avail_min setting has been changed to period_size*2 as far as alsa-lib is concerned, so of course two periods will always cause underruns. More than two work without certain underruns, but alsa-lib is sitting in a tight loop because it keeps calling poll() even though avail > avail_min. Anyone know what's going on here? Seems like a bug in alsa-lib, but I find it hard to believe no one would have noticed this before.

1 0

[alsa-devel] [PATCH 1/3] ASoC: wm8996: Handle failures to determine accessory polarity
by Mark Brown 26 Jan '12

26 Jan '12

If we get an indeterminate impedance with both headset polarities then give up and report the accessory as a headphone rather than continually retrying. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 38 ++++++++++++++++++++++++++------------ 1 files changed, 26 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index fc4f8fc..ab1fb0c 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -89,6 +89,7 @@ struct wm8996_priv { struct snd_soc_jack *jack; bool detecting; bool jack_mic; + int jack_flips; wm8996_polarity_fn polarity_cb; #ifdef CONFIG_GPIOLIB @@ -2427,6 +2428,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8996->jack = jack; wm8996->detecting = true; wm8996->polarity_cb = polarity_cb; + wm8996->jack_flips = 0; if (wm8996->polarity_cb) wm8996->polarity_cb(codec, 0); @@ -2542,6 +2544,19 @@ static void wm8996_hpdet_start(struct snd_soc_codec *codec) WM8996_HP_POLL, WM8996_HP_POLL); } +static void wm8996_report_headphone(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); + + /* Increase the detection rate a bit for responsiveness. */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 7 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2561,6 +2576,7 @@ static void wm8996_micd(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Jack removal detected\n"); wm8996->jack_mic = false; wm8996->detecting = true; + wm8996->jack_flips = 0; snd_soc_jack_report(wm8996->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2601,9 +2617,17 @@ static void wm8996_micd(struct snd_soc_codec *codec) /* If we detected a lower impedence during initial startup * then we probably have the wrong polarity, flip it. Don't * do this for the lowest impedences to speed up detection of - * plain headphones. + * plain headphones. If both polarities report a low + * impedence then give up and report headphones. */ if (wm8996->detecting && (val & 0x3f0)) { + wm8996->jack_flips++; + + if (wm8996->jack_flips > 1) { + wm8996_report_headphone(codec); + return; + } + reg = snd_soc_read(codec, WM8996_ACCESSORY_DETECT_MODE_2); reg ^= WM8996_HPOUT1FB_SRC | WM8996_MICD_SRC | WM8996_MICD_BIAS_SRC; @@ -2630,17 +2654,7 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm8996->detecting) { - dev_dbg(codec->dev, "Headphone detected\n"); - wm8996_hpdet_start(codec); - - /* Increase the detection rate a bit for - * responsiveness. - */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK | - WM8996_MICD_BIAS_STARTTIME_MASK, - 7 << WM8996_MICD_RATE_SHIFT | - 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); + wm8996_report_headphone(codec); } } } -- 1.7.9.rc1

1 2

[alsa-devel] [PATCH] ASoC: wm5100: Use pm_runtime for powerdown managment
by Mark Brown 26 Jan '12

26 Jan '12

Using pm_runtime to decide if the device should go into full power down has the dual advantage of allowing easier integration with non-DAPM reasons to power on the device (like the FLL) and allowing userspace to control the final power down which is useful for tuning retention of DSP firmware. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 116 ++++++++++++++++++++++----------------------- 1 files changed, 56 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 39de946..c6c3821 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -18,6 +18,7 @@ #include <linux/gcd.h> #include <linux/gpio.h> #include <linux/i2c.h> +#include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <linux/regulator/fixed.h> #include <linux/slab.h> @@ -1261,54 +1262,6 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_3_19, 1 }, }; -static int wm5100_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int ret; - - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - break; - - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - if (ret != 0) { - dev_err(codec->dev, - "Failed to enable supplies: %d\n", - ret); - return ret; - } - - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, - 1); - msleep(2); - } - - regcache_cache_only(wm5100->regmap, false); - regcache_sync(wm5100->regmap); - } - break; - - case SND_SOC_BIAS_OFF: - regcache_cache_only(wm5100->regmap, true); - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - break; - } - codec->dapm.bias_level = level; - - return 0; -} - static int wm5100_dai_to_base(struct snd_soc_dai *dai) { switch (dai->id) { @@ -1836,6 +1789,8 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (!Fout) { dev_dbg(codec->dev, "FLL%d disabled", fll_id); + if (fll->fout) + pm_runtime_put(codec->dev); fll->fout = 0; snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); return 0; @@ -1880,6 +1835,8 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Clear any pending completions */ try_wait_for_completion(&fll->lock); + pm_runtime_get_sync(codec->dev); + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, WM5100_FLL1_ENA); if (i2c->irq) @@ -1914,6 +1871,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } if (i == timeout) { dev_err(codec->dev, "FLL%d lock timed out\n", fll_id); + pm_runtime_put(codec->dev); return -ETIMEDOUT; } @@ -2377,9 +2335,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(wm5100->regmap, true); - - for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, WM5100_OUT_VU); @@ -2405,14 +2360,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) } } - /* We'll get woken up again when the system has something useful - * for us to do. - */ - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - return 0; err_gpio: @@ -2444,7 +2391,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, - .set_bias_level = wm5100_set_bias_level, .idle_bias_off = 1, .reg_cache_size = WM5100_MAX_REGISTER, .volatile_register = wm5100_soc_volatile, @@ -2661,6 +2607,10 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, } } + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); @@ -2714,6 +2664,51 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm5100_runtime_suspend(struct device *dev) +{ + struct wm5100_priv *wm5100 = dev_get_drvdata(dev); + + regcache_cache_only(wm5100->regmap, true); + regcache_mark_dirty(wm5100->regmap); + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return 0; +} + +static int wm5100_runtime_resume(struct device *dev) +{ + struct wm5100_priv *wm5100 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 1); + msleep(2); + } + + regcache_cache_only(wm5100->regmap, false); + regcache_sync(wm5100->regmap); + + return 0; +} +#endif + +static struct dev_pm_ops wm5100_pm = { + SET_RUNTIME_PM_OPS(wm5100_runtime_suspend, wm5100_runtime_resume, + NULL) +}; + static const struct i2c_device_id wm5100_i2c_id[] = { { "wm5100", 0 }, { } @@ -2724,6 +2719,7 @@ static struct i2c_driver wm5100_i2c_driver = { .driver = { .name = "wm5100", .owner = THIS_MODULE, + .pm = &wm5100_pm, }, .probe = wm5100_i2c_probe, .remove = __devexit_p(wm5100_i2c_remove), -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: wm5100: Convert to devm_regulator_bulk_get()
by Mark Brown 26 Jan '12

26 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 12 ++++-------- 1 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index c1c8bdb..39de946 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2504,8 +2504,9 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2517,7 +2518,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_core; + goto err_regmap; } if (wm5100->pdata.ldo_ena) { @@ -2686,9 +2687,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_core: - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); err_regmap: regmap_exit(wm5100->regmap); err: @@ -2711,8 +2709,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); regmap_exit(wm5100->regmap); return 0; -- 1.7.9.rc1

1 0