November 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: add codec support for mini6410 and real6410 boards
by Paulius Zaleckas 18 Nov '10

18 Nov '10

Signed-off-by: Paulius Zaleckas <paulius.zaleckas(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8a6b53c..b1fcbd3 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -132,7 +132,7 @@ config SND_S3C24XX_SOC_RX1950_UDA1380 config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_MINI6410 || MACH_REAL6410) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help

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[alsa-devel] ASoC: Cyclic dependency between soc.h and soc-dapm.h
by Lars-Peter Clausen 17 Nov '10

17 Nov '10

The commit "ASoC: Decouple DAPM from CODECs" introduced a cyclic dependency between soc.h and soc-dapm.h which leads to the following compile error, if a driver includes soc-dapm.h without having included soc.h before: In file included from include/sound/soc-dapm.h:19, from sound/soc/codecs/jz4740.c:25: include/sound/soc.h:484: error: field 'dapm' has incomplete type If soc.h has been included before the error does not happen, but then soc-dapm.h already got included trough soc.h, so the driver does not really have to include it again. So I wonder if soc-dapm.h should be made completely internal and not be included anywhere else but soc.h anymore? Or if the cyclic dependency should be broken somehow, maybe by introducing a third header file? - Lars

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[alsa-devel] [PATCH] ASoC: davinci: fixes for multi-component
by Chris Paulson-Ellis 17 Nov '10

17 Nov '10

Multi-component commit f0fba2ad broke a few things which this patch should fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be good if those with access to other Davinci boards could test. -- The multi-component commit put the initialisation of snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c). The initialisation had to be moved from the probe function in these drivers because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver. Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in davinci-pcm.c) before hw_params is called. I have moved the initialisation to a new snd_soc_dai_ops.startup function in each of these drivers. This fix indicates that all platforms that use davinci-pcm must have been broken and need to test with this fix. -- The multi-component commit also changed the McBSP driver name from "davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board level references to the driver name. This change is understandable, as there is a similarly named "davinci-mcasp" driver in davinci-mcasp.c. There is probably no 'correct' name for this driver. The DM6446 datasheet calls it the "ASP" and describes it as a "specialised McBSP". The DM355 datasheet calls it the "ASP" and describes it as a "specialised ASP". The DM365 datasheet calls it the "McBSP". Rather than fix this problem by reverting to "davinci-asp", I've elected to avoid future confusion with the "davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also consistent with the names of the functions in the driver. There are other fixes required, so it was never going to be as simple as a revert anyway. -- The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has 2 ports), so I've changed the the id of the platform_device from 0 to -1. -- In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link structure with the DM355 EVM as they use different cpu DAI names (the DM355 has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port). This also means that the 2 boards need different snd_soc_card structures. -- The codec_name entries in davinci-evm.c didn't match the i2c ids in the board files. I have only checked and fixed the details of the names used for the McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should check the others. Signed-off-by: Chris Paulson-Ellis <chris(a)edesix.com> --- arch/arm/mach-davinci/dm355.c | 6 ++-- arch/arm/mach-davinci/dm365.c | 6 ++-- arch/arm/mach-davinci/dm644x.c | 4 +- sound/soc/davinci/davinci-evm.c | 40 ++++++++++++++++++++++++++--------- sound/soc/davinci/davinci-i2s.c | 15 +++++++++--- sound/soc/davinci/davinci-mcasp.c | 13 +++++++++-- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 13 +++++++++-- 8 files changed, 69 insertions(+), 30 deletions(-) diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index 9be261b..2652af1 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -359,8 +359,8 @@ static struct clk_lookup dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("davinci-asp.0", NULL, &asp0_clk), - CLK("davinci-asp.1", NULL, &asp1_clk), + CLK("davinci-mcbsp.0", NULL, &asp0_clk), + CLK("davinci-mcbsp.1", NULL, &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK("spi_davinci.0", NULL, &spi0_clk), @@ -664,7 +664,7 @@ static struct resource dm355_asp1_resources[] = { }; static struct platform_device dm355_asp1_device = { - .name = "davinci-asp", + .name = "davinci-mcbsp", .id = 1, .num_resources = ARRAY_SIZE(dm355_asp1_resources), .resource = dm355_asp1_resources, diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index a12065e..c466d71 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -459,7 +459,7 @@ static struct clk_lookup dm365_clks[] = { CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK("davinci_voicecodec", NULL, &voicecodec_clk), - CLK("davinci-asp.0", NULL, &asp0_clk), + CLK("davinci-mcbsp", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), CLK(NULL, NULL, NULL), @@ -922,8 +922,8 @@ static struct resource dm365_asp_resources[] = { }; static struct platform_device dm365_asp_device = { - .name = "davinci-asp", - .id = 0, + .name = "davinci-mcbsp", + .id = -1, .num_resources = ARRAY_SIZE(dm365_asp_resources), .resource = dm365_asp_resources, }; diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 0608dd7..9a2376b 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -302,7 +302,7 @@ static struct clk_lookup dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("davinci-asp", NULL, &asp_clk), + CLK("davinci-mcbsp", NULL, &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -580,7 +580,7 @@ static struct resource dm644x_asp_resources[] = { }; static struct platform_device dm644x_asp_device = { - .name = "davinci-asp", + .name = "davinci-mcbsp", .id = -1, .num_resources = ARRAY_SIZE(dm644x_asp_resources), .resource = dm644x_asp_resources, diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index a2cf64b..fd96905 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -158,12 +158,23 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) } /* davinci-evm digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link evm_dai = { +static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", - .cpu_dai_name = "davinci-mcasp.0", + .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-001b", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link dm355_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcbsp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-001b", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, @@ -173,10 +184,10 @@ static struct snd_soc_dai_link dm365_evm_dai = { #ifdef CONFIG_SND_DM365_AIC3X_CODEC .name = "TLV320AIC3X", .stream_name = "AIC3X", - .cpu_dai_name = "davinci-i2s", + .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .init = evm_aic3x_init, - .codec_name = "tlv320aic3x-codec.0-001a", + .codec_name = "tlv320aic3x-codec.1-0018", .ops = &evm_ops, #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", @@ -220,10 +231,17 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 evm audio machine driver */ -static struct snd_soc_card snd_soc_card_evm = { - .name = "DaVinci EVM", - .dai_link = &evm_dai, +/* davinci dm6446 evm audio machine driver */ +static struct snd_soc_card dm6446_snd_soc_card_evm = { + .name = "DaVinci DM6446 EVM", + .dai_link = &dm6446_evm_dai, + .num_links = 1, +}; + +/* davinci dm355 evm audio machine driver */ +static struct snd_soc_card dm355_snd_soc_card_evm = { + .name = "DaVinci DM355 EVM", + .dai_link = &dm355_evm_dai, .num_links = 1, }; @@ -262,10 +280,10 @@ static int __init evm_init(void) int ret; if (machine_is_davinci_evm()) { - evm_snd_dev_data = &snd_soc_card_evm; + evm_snd_dev_data = &dm6446_snd_soc_card_evm; index = 0; } else if (machine_is_davinci_dm355_evm()) { - evm_snd_dev_data = &snd_soc_card_evm; + evm_snd_dev_data = &dm355_snd_soc_card_evm; index = 1; } else if (machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &dm365_snd_soc_card_evm; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d46b545..9e0e565 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -426,9 +426,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t fmt; unsigned element_cnt = 1; - dai->capture_dma_data = dev->dma_params; - dai->playback_dma_data = dev->dma_params; - /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -601,6 +598,15 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -612,6 +618,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -749,7 +756,7 @@ static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .driver = { - .name = "davinci-i2s", + .name = "davinci-mcbsp", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 86918ee..fb55d2c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -715,9 +715,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int word_length; u8 fifo_level; - cpu_dai->capture_dma_data = dev->dma_params; - cpu_dai->playback_dma_data = dev->dma_params; - davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; @@ -799,7 +796,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 009b652..6c6666a 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -84,7 +84,7 @@ static struct snd_soc_ops sffsdr_ops = { static struct snd_soc_dai_link sffsdr_dai = { .name = "PCM3008", /* Codec name */ .stream_name = "PCM3008 HiFi", - .cpu_dai_name = "davinci-asp.0", + .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "pcm3008-hifi", .codec_name = "pcm3008-codec", .platform_name = "davinci-pcm-audio", diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index ea232f6..fb4cc1e 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -97,9 +97,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, &davinci_vcif_dev->dma_params[substream->stream]; u32 w; - dai->capture_dma_data = davinci_vcif_dev->dma_params; - dai->playback_dma_data = davinci_vcif_dev->dma_params; - /* Restart the codec before setup */ davinci_vcif_stop(substream); davinci_vcif_start(substream); @@ -174,9 +171,19 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int davinci_vcif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); + return 0; +} + #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; -- 1.7.2.3

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[PATCH 3/3] ALSA: HDA: Add Front control for ALC887
by David Henningsson 17 Nov '10

17 Nov '10

HP and Line out share DAC, so we need a switch to turn line out off. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4cc7bb2..86df37c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10836,8 +10836,16 @@ static int alc882_parse_auto_config(struct hda_codec *codec) err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); if (err < 0) return err; - if (codec->vendor_id == 0x10ec0887) + if (codec->vendor_id == 0x10ec0887) { err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + /* We need something to switch that doesn't affect headphones */ + if (spec->autocfg.line_out_pins[0]) + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, "Front", + HDA_COMPOSE_AMP_VAL(spec->autocfg.line_out_pins[0], 3, 0, HDA_OUTPUT)); + + } else err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) -- 1.7.1 --------------050400000103080707060807--

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[alsa-devel] Sound card OpenSuse 11.3
by Luca Samiolo 17 Nov '10

17 Nov '10

Good morning, I'm Luca. I want to ask information: can I get a list of sound cards compatible with OpenSuse 10.3? Thank you Luca Samiolo IT Support & Help Desk luca.samiolo(a)nemes.it ------------------------------------------------------------------------------------------------------------------------------ The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer.

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[alsa-devel] [PATCH] ASoC: Fix register cache setup WM8994 for multi-component
by Mark Brown 16 Nov '10

16 Nov '10

During the multi-component conversion the WM8994 register cache init got lost. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 363b3ba..f598ca0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4342,6 +4342,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm8994); + codec->reg_cache = &wm8994->reg_cache; + wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; -- 1.7.1

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[alsa-devel] nvidia initial sound being skipped
by David Shirley 16 Nov '10

16 Nov '10

Hi Developers, I am using an nvidia MCP78 to connect to a Denon 1910 AVR via HDMI. I was using Kernel 2.6.32.21 and noticed that the initial few seconds of my songs was being chopped off. After seeing http://www.spinics.net/lists/alsa-devel/msg29408.html I upgraded to 2.6.35.8 (which has that fix in it) and the problem was been reduced considerably. But it is still occurring to some extent. So to see if the HDMI was the problem I changed it to coax spdif - no change/same problem. Now I think it could be mplayer, so I try audacious - no change/same problem. Now I think it could be the flac file/decoding, so I decode the flac to wav and use aplay - no change/same problem. The ONLY way I can get the first 0.5seconds of the file to play is by using audacious and then MANUALLY rewinding (by dragging the scroll bar thingy) back to the start... then it will play (on either flac or wav) Co-incidentally when I do this I get the following error message dumped to console: alsa: snd_pcm_prepare failed: Device or resource busy. I hope someone can help :) I am happy to get debug or whatever - just let me know what you need! mythtv@crystal:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 lspci -vvvv 00:07.0 Audio device: nVidia Corporation MCP72XE/MCP72P/MCP78U/MCP78S High Definition Audio (rev a1) Subsystem: ASUSTeK Computer Inc. M3N72-D Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 11 Region 0: Memory at fe020000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel Cheers David

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[alsa-devel] SEGFAULT : snd_pcm_mmap_begin/commit
by Irfan Shaikh 16 Nov '10

16 Nov '10

Hello All, I have been facing dificulty in using snd_pcm_mmap_begin and snd_pcm_mmap_commit. Shows Seg fault as pointed by comment in code below. Please tell me what wrong i am doing and how do i correct. /###################################################################################/ unsigned int count = (alsaParams->frames/(((alsaParams->bit_per_sample)*(alsaParams->num_channel)) / 8)); unsigned char *data = alsaParams->data;https://owa.sasken.com/owa/?ae=Item&t=IPM.Note&a=New# snd_pcm_sframes_t frames_available, frames_transmit; const snd_pcm_channel_area_t *area; snd_pcm_uframes_t offset; char *outbuffer; int result; /*available space in audio-buffer to write to*/ frames_available = snd_pcm_avail_update(alsaParams->pcm_handle); /* * prepare the areas * and get back area, offset and frames_transmit * frames_transmit gives back the REAL value of available frames which could be written to in the mmaped-area * frames_transmit are less than frames_available */ frames_transmit = frames_available; snd_pcm_mmap_begin(alsaParams->pcm_handle, &area, &offset, &frames_transmit); /*calculates the mmaped-buffer which could be written to*/ outbuffer = ((char *) area->addr + (area->first + area->step * offset)/ 8); // ****** SEGFAULT ******* /*gives back the virtually written (writable) frames which should be always == frames_transmit*/ result = snd_pcm_mmap_commit(alsaParams->pcm_handle, offset, frames_transmit); memcpy(outbuffer, data, count); //data is actual data I think if above code works fine should play sound. /###################################################################################/ SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[PATCH 2/3] ALSA: HDA: Rename "Front" to "Master Front" on ALC861VD and ALC887
by David Henningsson 16 Nov '10

16 Nov '10

Since "Front" controls headphones as well on these two codecs, they should be labeled "Master Front" in order to be handled correct by PulseAudio. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce4017e..4cc7bb2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16966,7 +16966,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"}; + static const char *chname[4] = {"Master Front", "Surround", "CLFE", "Side"}; hda_nid_t nid_v, nid_s; int i, err; -- 1.7.1 --------------040808030805090205070405--

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[alsa-devel] Direct write (using mmap_bigin and commit)
by Irfan Shaikh 16 Nov '10

16 Nov '10

Hello, I am reading an raw PCM file and want to write on PCM device using direct write (interleaved). I have gone through PCM.c test application. I am trying tio write PCM audio data as follows. I am unable to use snd_pcm_mmap_begin and snd_pcm_mmap_commit properly. Please help me to use using these APIs Question : 1) How can i write to Ring buffer area (any pcm audio data)? //Any example or explanation other than pcm.c 2) Do i need to write diffferently for each channel as shown in pcm.c I am trying to do following but not succeded (reading pcm file and writing content on device) . ANy thing played on audio device would help me in understanding. snd_pcm_channel_area_t my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; int result; char *buffer; while(1) { avail = snd_pcm_avail_update(handle); if(avail>=period_size) { size=periodsize while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); mybuffer=(char *)malloc(frames*2); result = fread (mybuffer,1,(frames*2),pFile); my_areas.addr=mybuffer; // IS THIS CORRECT ? WILL THIS WORK ? How can i write my audio file data here in areas? commitres = snd_pcm_mmap_commit(handle, offset, frames); } } SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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