March 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: soc-core: use snd_soc_register_dais() on codec
by Kuninori Morimoto 02 Mar '13

02 Mar '13

snd_soc_register_dais() considers dai counts inside. snd_soc_register_codec() does not need to care for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ceac893..5f44d22 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4047,11 +4047,10 @@ int snd_soc_register_codec(struct device *dev, mutex_unlock(&client_mutex); /* register any DAIs */ - if (num_dai) { - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) - dev_err(codec->dev, "ASoC: Failed to regster" - " DAIs: %d\n", ret); + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto fail_codec_name; } dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); @@ -4073,7 +4072,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); void snd_soc_unregister_codec(struct device *dev) { struct snd_soc_codec *codec; - int i; list_for_each_entry(codec, &codec_list, list) { if (dev == codec->dev) @@ -4082,9 +4080,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - if (codec->num_dai) - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(dev, codec->num_dai); mutex_lock(&client_mutex); list_del(&codec->list); -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/5 v7][RFC] ASoC: add snd_soc_register_cpu()
by Mark Brown 02 Mar '13

02 Mar '13

On Fri, Mar 01, 2013 at 09:42:10PM +0000, Liam Girdwood wrote: > On Fri, 2013-03-01 at 21:50 +0100, Lars-Peter Clausen wrote: > > object. I mean the distinction is mostly for historic reasons from a time where > > it was only possible to bind a the cpu_dai of a link to a CPU DAI and the > > codec_dai to a CODEC DAI. But these dais both can for example be bound CODEC > > DAIs. I think to have a common base here would allow us to do some cleanups to > > the ASoC core and also avoid the duplication with snd_soc_of_get_cpu_dai_name > > and snd_soc_of_get_codec_dai_name which was introduced in this series. > I've been thinking about this distinction for some time too (since > multi-component support). You are right, the division with CODEC and CPU > DAIs is entirely historical. The same logic also applies to platform and > codec drivers too. The DAIs are now the same of course, it's just the containing object that differs. Platform drivers are a bit different to CODEC drivers for me though in that they encapsulate the DMA equivalent of a DAI - I can see a SoC having multiple platform drivers if it's got a couple of different DMA controllers for example (I've seen some which have a fancier one on faster buses for example). > This does need some mechanical cleanup where we have a generic audio > component driver object that contains generic DAIs. i.e. codec and > platform drivers would become generic component drivers with generic > DAIs. There's also the perennial questions about code churn and if it's worth it of course. Should we be adding a chip driver type rather than a CPU one here and then going through and doing the work to make sure that you can use a chip driver wherever you use a CODEC driver so we can start converting things over?

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Re: [alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by wendelin klimann 01 Mar '13

01 Mar '13

Hello and thanks for the support I would advice to update. Eventually you want to upstream the codec driver, > so > it is going to be easier for you. > Well i followed your advice and updated to kernel version 3.7.4+ and to ALSA version k3.7.4+, and i will be glad to upstram the finalized driver. As you recommended i am using your omap_twl4030.c file as a starting-point for my driver development, and i am trying to get the adapted machine driver to run but without success. The machine driver is not even going into * static __devinit int omap_pcm3168_probe(struct platform_device *pdev)* and i am really stuck at this point. It would be really nice if you could help me. Regards Wendelin File: /* * omap-pcm3168.c -- PCM3168 ASoC driver for the BeagleBoard. * * based on: * > omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec * > * > Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com * > All rights reserved. * > * > Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA * */ #include <linux/platform_device.h> //#include <linux/platform_data/omap-twl4030.h> #include <linux/module.h> #include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include "omap-mcbsp.h" #include "omap-pcm.h" static int omap_pcm3168_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; unsigned int fmt; int ret; fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF; /* Set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { dev_err(card->dev, "can't set codec DAI configuration\n"); return ret; } /* Set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { dev_err(card->dev, "can't set cpu DAI configuration\n"); return ret; } return 0; } static struct snd_soc_ops omap_pcm3168_ops = { .hw_params = omap_pcm3168_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_pcm3168_dai_links[] = { { .name = "PCM3168", .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "pcm3168-hifi", .platform_name = "omap-pcm-audio", .codec_name = "pcm3168-codec.0", .ops = &omap_pcm3168_ops, }, }; /* Audio machine driver */ static struct snd_soc_card omap_pcm3168_card = { .owner = THIS_MODULE, .dai_link = omap_pcm3168_dai_links, .num_links = ARRAY_SIZE(omap_pcm3168_dai_links), }; static __devinit int omap_pcm3168_probe(struct platform_device *pdev) { printk("in __devinit omap_pcm3168_probe -> by kli"); struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_pcm3168_card; int ret = 0; card->dev = &pdev->dev; if (node) { struct device_node *dai_node; if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } dai_node = of_parse_phandle(node, "ti,mcbsp", 0); if (!dai_node) { dev_err(&pdev->dev, "McBSP node is not provided\n"); return -EINVAL; } omap_pcm3168_dai_links[0].cpu_dai_name = NULL; omap_pcm3168_dai_links[0].cpu_of_node = dai_node; } /*else if (pdata) { if (pdata->card_name) { card->name = pdata->card_name; } else { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; } }*/ else { dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } return 0; } static int __devexit omap_pcm3168_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static const struct of_device_id omap_pcm3168_of_match[] = { {.compatible = "ti,omap-pcm3168", }, { }, }; MODULE_DEVICE_TABLE(of, omap_pcm3168_of_match); static struct platform_driver omap_pcm3168_driver = { .driver = { .name = "omap-pcm3168", .owner = THIS_MODULE, // .pm = &snd_soc_pm_ops, // .of_match_table = omap_pcm3168_of_match, }, .probe = omap_pcm3168_probe, .remove = __devexit_p(omap_pcm3168_remove), }; module_platform_driver(omap_pcm3168_driver); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("ALSA SoC PCM3168 add on Soundcard"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:omap-pcm3168");

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[alsa-devel] Thinkpad T400 - internal mic doesn't work, can't set CAPTURE - schematics included
by Arkadiusz Miśkiewicz 01 Mar '13

01 Mar '13

Hi, My Thinkpad T400 (2764CTO model) doesn't have ability to set CAPTURE on microphone inputs with alsa. Most likely some quirk is missing. The laptop has internal microphone, microphone in input, headphones out output and 2 internal speakers. That's all. Is uses Conexant CX20561_15Z internally and there is detailed T400 schematic available: http://kythuatphancung.com/download/lenovo-thinkpad-t400-schematic- diagram.html?dl (for audio look around page 43). Kernels 3.7.x, fresh 3.8 - no luck. Tried even linux-next from today but it dies early on alsa unrelated issues. I've tried model options laptop, hp, hp-dv6736, hp-700, toshiba, ideapad, auto. None of these work fine. With model=hp-700 I can set CAPTURE but it doesn't work (silent audio is captured using arecord or in google hangout). External mic connected also doesn't work with these setups. http://www.alsa-project.org/db/?f=394ff4cdca33e75c8a47203d42b89d8de36deca7 [3.8 kernel] http://www.alsa-project.org/db/?f=302877e50f99835075d038207f6db0dfd89873b0 [3.7 kernel] Thinkpad is 2764CTO model exactly. I can test various hacks, patches etc easily. Could someone look at schematic and provide quirks to test? Thanks! -- Arkadiusz Miśkiewicz, arekm / maven.pl

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[alsa-devel] [PATCH v3] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 01 Mar '13

01 Mar '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. For TDM mode, McASP is capable to play or receive 32 channels for one serializer. McAsp has support for max 16 serializer, therefore max channels is 32 * 8. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 66 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 +++++---- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 68 insertions(+), 15 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d2ca682..0a1af1a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, @@ -1015,8 +1057,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->op_mode = val; ret = of_property_read_u32(np, "tdm-slots", &val); - if (ret >= 0) + if (ret >= 0) { + if (val < 2 || val > 32) { + dev_err(&pdev->dev, + "tdm-slots must be in rage [2-32]\n"); + ret = -EINVAL; + goto nodata; + } + pdata->tdm_slots = val; + } ret = of_property_read_u32(np, "num-serializer", &val); if (ret >= 0) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: wm5102: Correct OUT2 volume and switch names
by Charles Keepax 01 Mar '13

01 Mar '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ab69c83..deb1097 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -755,7 +755,7 @@ SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), @@ -767,7 +767,7 @@ SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, +SOC_DOUBLE_R_TLV("HPOUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: arizona: Don't enable FLL on REFCLK configuration
by Charles Keepax 01 Mar '13

01 Mar '13

Enabling the FLL when REFCLK is being configured is not what the user would expect and can cause issues if SYNCCLK has no specified frequency. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 15 ++++++--------- 1 files changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6837863..eb25ab9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1148,17 +1148,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (source < 0) return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) + if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (fll->fout) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); if (ret != 0) return ret; if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, Fout); + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, + fll->fout); if (ret != 0) return ret; } @@ -1166,12 +1166,9 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - fll->fout = Fout; - if (Fout) { + if (fll->fout) { arizona_enable_fll(fll, &ref, &sync); - } else { - arizona_disable_fll(fll); } return 0; -- 1.7.2.5

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[alsa-devel] [PATCH] hdajackretask: Fix "advanced override" bug
by David Henningsson 01 Mar '13

01 Mar '13

The values were not correctly retreived from the comboboxes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/main-gtk.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/hdajackretask/main-gtk.c b/hdajackretask/main-gtk.c index 65d0f3a..4227669 100644 --- a/hdajackretask/main-gtk.c +++ b/hdajackretask/main-gtk.c @@ -183,14 +183,14 @@ static void update_user_pin_config(ui_data_t* ui) continue; if (ui->free_overrides) { - int i; + int j; int index; unsigned long val = 0; - for (i = 0; i < FREE_OVERRIDES_COUNT; i++) { - index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->jacktype)); - if (index < 0) + for (j = 0; j < FREE_OVERRIDES_COUNT; j++) { + index = gtk_combo_box_get_active(GTK_COMBO_BOX(pin_ui->free_override_cb[j])); + if (index < 0) break; - val += get_free_override_list(i)[index].value; + val += get_free_override_list(j)[index].value; } if (index < 0) continue; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: snd-core: tidyup snd_soc_register_codec() fail case
by Kuninori Morimoto 01 Mar '13

01 Mar '13

kfree() on snd_soc_register_codec() was summarized to one place. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 689eb04..ceac893 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3983,8 +3983,8 @@ int snd_soc_register_codec(struct device *dev, /* create CODEC component name */ codec->name = fmt_single_name(dev, &codec->id); if (codec->name == NULL) { - kfree(codec); - return -ENOMEM; + ret = -ENOMEM; + goto fail_codec; } if (codec_drv->compress_type) @@ -4023,7 +4023,7 @@ int snd_soc_register_codec(struct device *dev, reg_size, GFP_KERNEL); if (!codec->reg_def_copy) { ret = -ENOMEM; - goto fail; + goto fail_codec_name; } } } @@ -4057,8 +4057,9 @@ int snd_soc_register_codec(struct device *dev, dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name); return 0; -fail: +fail_codec_name: kfree(codec->name); +fail_codec: kfree(codec); return ret; } -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: wm8960: Correct register 0 and 1 defaults
by Mark Brown 01 Mar '13

01 Mar '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb9273..4cb49eb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -53,8 +53,8 @@ * using 2 wire for device control, so we cache them instead. */ static const struct reg_default wm8960_reg_defaults[] = { - { 0x0, 0x0097 }, - { 0x1, 0x0097 }, + { 0x0, 0x00a7 }, + { 0x1, 0x00a7 }, { 0x2, 0x0000 }, { 0x3, 0x0000 }, { 0x4, 0x0000 }, -- 1.7.10.4

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