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

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Use driver_data field of struct i2c_device_id to identify models
by Axel Lin 30 Sep '11

30 Sep '11

Save model information in driver_data so we can simplify the implementation. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 13 ++++--------- 1 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0963c4c..597abb9 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1495,9 +1495,9 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { */ static const struct i2c_device_id aic3x_i2c_id[] = { - [AIC3X_MODEL_3X] = { "tlv320aic3x", 0 }, - [AIC3X_MODEL_33] = { "tlv320aic33", 0 }, - [AIC3X_MODEL_3007] = { "tlv320aic3007", 0 }, + { "tlv320aic3x", AIC3X_MODEL_3X }, + { "tlv320aic33", AIC3X_MODEL_33 }, + { "tlv320aic3007", AIC3X_MODEL_3007 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1512,7 +1512,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_pdata *pdata = i2c->dev.platform_data; struct aic3x_priv *aic3x; int ret; - const struct i2c_device_id *tbl; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1531,11 +1530,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->gpio_reset = -1; } - for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) { - if (!strcmp(tbl->name, id->name)) - break; - } - aic3x->model = tbl - aic3x_i2c_id; + aic3x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); -- 1.7.4.1

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[alsa-devel] [PATCH 1/2] ASoC: Add BUG() assertion if max98088_get_channel returns -EINVAL
by Axel Lin 30 Sep '11

30 Sep '11

The callers use the return value of max98088_get_channel as array index to access max98088->dai[] array. Add BUG() assertion for out of bound access of max98088->dai[] array. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/max98088.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ac65a2d..aaca91c 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1703,6 +1703,7 @@ static int max98088_get_channel(const char *name) return 0; if (strcmp(name, "EQ2 Mode") == 0) return 1; + BUG(); return -EINVAL; } -- 1.7.4.1

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[alsa-devel] [PATCH v3 0/4] ASoC: twl6040: Gain ramp code cleanups
by Peter Ujfalusi 30 Sep '11

30 Sep '11

Hello, Changes since v2: - Use cancel_delayed_work_sync to make sure that we do not have pending work, or executing work when modifying the variables used by the ramp code. Intro mail from v2: Changes since v1: - Loop counters cleaned, commented in patch 2 - Use cancel_delayed_work_sync to make sure that we do not have pending work, or executing work when modifying the variables used by the ramp code. - patches already taken left out. Intro from v1: the following series cleans up the gain ramp code found in the twl6040 codec driver. Main changes: - use one workqueue for the twl6040 codec driver (instead of the original 3) - Delays between the steps does not need to be different among the range. I assume, that the original code contained copy-paste snippets from wm8350 for this part - Cleanups for the DAPM_OUT_DRV_E event handler code. - delayed_works moved within their corresponding struct. The series has been generated on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch + ASoC: omap-mcpdm/twl6040: Offset cancellation series. Side note: I have done this to better understand (while cleaning up the twl6040 driver) the requirements for the ramp code, and to study the possibility of adding support in the core for this (to handle the wm8350, and twl6040 in a generic way later). I'm still looking at the optimal implementation without ending up with too complicated code/structures... Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: One workqueue should be enough ASoC: twl6040: correct loop counters for HS/HF ramp code ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event ASoC: twl6040: Simplify code in out_drv_event for pending work check sound/soc/codecs/twl6040.c | 81 +++++++++++++++++-------------------------- 1 files changed, 32 insertions(+), 49 deletions(-) -- 1.7.6.1

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[alsa-devel] [PATCH for-3.2 1/2] ASoC: omap-mcbsp: Fix FS polarity for LEFT_J, DSP_A and DSP_B formats
by Jarkko Nikula 30 Sep '11

30 Sep '11

Commit 75d9ac4 ("ASoC: Allow DAI formats to be specified in the dai_link") changed DAI format flag values and we cannot simply invert anymore e.g. frame-sync with ^= SND_SOC_DAIFMT_NB_IF (which was anyway misuse) as there is no anymore fixed bit position for bit-clock or frame-sync inversion. Fix this by relying only on DAI format flag values passed to us and by not making any assumption on individual bit positions. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1391ea0..894f2f3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -398,7 +398,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; + bool inv_fs = false; if (mcbsp_data->configured) return 0; @@ -430,21 +430,21 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->xcr2 |= XDATDLY(0); regs->spcr1 |= RJUST(2); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); /* Invert FS polarity configuration */ - temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + inv_fs = true; break; default: /* Unsupported data format */ @@ -468,7 +468,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. @@ -489,6 +489,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } + if (inv_fs == true) + regs->pcr0 ^= FSXP | FSRP; return 0; } -- 1.7.6.3

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[alsa-devel] [PATCH resend] ASoC: sdp4430: Fix string for FM input name
by Peter Ujfalusi 30 Sep '11

30 Sep '11

From: "Ujfalusi, Peter" <peter.ujfalusi(a)ti.com> The name contains invalid valid character (/), which causes problems when trying to create the debugfs directory structure: ASoC: Failed to create Aux/FM Stereo In debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- Hi Mark, Liam, Could you take this patch, it has been not taken for upstream. Thanks, Peter sound/soc/omap/sdp4430.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 249d84b..79ec76d 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,8 +113,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- 1.7.6.1

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[alsa-devel] [PATCH] ASoC: sn95031: Do not use static variable for channel_index
by Axel Lin 30 Sep '11

30 Sep '11

No reason to use static variable for channel_index. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sn95031.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index adb85bf..02a4857 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -103,7 +103,7 @@ static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) { int base_addr, chnl_addr; int value; - static int channel_index; + int channel_index; /* Index of the first channel in which the stop bit is set */ channel_index = find_free_channel(sn95031_codec); -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Add missed regulator_unregister_notifier and regulator_bulk_free in wm8995_remove
by Axel Lin 30 Sep '11

30 Sep '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8995.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 74ae599..e05ee79 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1573,9 +1573,16 @@ static int wm8995_resume(struct snd_soc_codec *codec) static int wm8995_remove(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; + int i; wm8995 = snd_soc_codec_get_drvdata(codec); wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) + regulator_unregister_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); return 0; } -- 1.7.4.1

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[alsa-devel] [PATCH 1/2] ASoC: Remove unused "control_data" field of struct cs4271_private
by Axel Lin 30 Sep '11

30 Sep '11

The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs4271.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 083aab9..23d1bd5 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -156,7 +156,6 @@ static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ enum snd_soc_control_type bus_type; - void *control_data; unsigned int mclk; bool master; bool deemph; @@ -466,8 +465,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; - codec->control_data = cs4271->control_data; - if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -555,7 +552,6 @@ static int __devinit cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->control_data = spi; cs4271->bus_type = SND_SOC_SPI; return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, @@ -595,7 +591,6 @@ static int __devinit cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->control_data = client; cs4271->bus_type = SND_SOC_I2C; return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, -- 1.7.4.1

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[alsa-devel] VT1708 clicks & pops
by Forest Bond 30 Sep '11

30 Sep '11

Hi, I have two boards with a VT1708: * VIA EPIA EX15000G * VIA VB8002 I'm using VLC to play audio via PulseAudio to the analog outputs. The VB8002 sounds great, but the EX15000G has random pops and clicks in the output. Both machines are running exactly the same configuration (they are using the same pre-built OS image). I've tested on the EX15000G using a few different driver versions. One is essentially the version from Ubuntu 10.04, but rebuilt due to unrelated kernel changes. The other is alsa-driver snapshot alsa-driver-1.0.24.72.g7c48f.1314.geb1d0. You can find output from alsa-info.sh as follows: EX15000G (using Ubuntu 10.04 driver): http://www.alsa-project.org/db/?f=ea56226a7917391d6adf9078650d3a9395f51b17 EX15000G (using driver from alsa-driver-1.0.24.72.g7c48f.1314.geb1d0): http://www.alsa-project.org/db/?f=d0265991ec3e975820dc439470aea1df925ca1db VB8002 (using Ubuntu 10.04 driver): http://www.alsa-project.org/db/?f=611cc772a3fab7552bbda193b6ad7998f66a60fb Any ideas what the problem might be? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

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[alsa-devel] PATCH: [plug] Move fast_ops copy after slave configuration.
by Reilly Grant 30 Sep '11

30 Sep '11

The plug plugin copies the fast_ops structure and fast_op_arg from its slave PCM on initialization and after inserting a new plugin. This works unless the slave PCM is also an instance of the plug plugin because plug can change its fast_ops structure during _snd_pcm_hw_params. Instead of copying fast_ops in snd_pcm_plug_insert_plugins wait until the end of snd_pcm_plug_hw_params when the slave has been fully initialized. This fixes a crash when two instances of plug are instantiated back to back and the format conversion plugin inserted by the second is skipped by the first because the fast_ops pointer is not properly updated. --- src/pcm/pcm_plug.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..72456d5 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -659,8 +659,6 @@ static int snd_pcm_plug_insert_plugins(snd_pcm_t *pcm, } if (err) { plug->gen.slave = new; - pcm->fast_ops = new->fast_ops; - pcm->fast_op_arg = new->fast_op_arg; } k++; } @@ -1049,6 +1047,9 @@ static int snd_pcm_plug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); snd_pcm_unlink_appl_ptr(pcm, plug->req_slave); + + pcm->fast_ops = slave->fast_ops; + pcm->fast_op_arg = slave->fast_op_arg; snd_pcm_link_hw_ptr(pcm, slave); snd_pcm_link_appl_ptr(pcm, slave); return 0; -- 1.7.3.4

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