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

[alsa-devel] ASoC: multi-component updates from RFC
by Liam Girdwood 22 Jul '10

22 Jul '10

I've just pushed some updates to multi-component that came out of the RFC discussion. The main update is the use of dev_name (or DAI name for devices with multiple DAIs) to identify ASoC components. This replaces the current method of using a struct codec|dai|platform pointer and ID to define a DAI link. e.g. we can now define a DAI link as follows :- static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", .cpu_dai_name = "omap-mcbsp-dai.1", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }, { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp-dai.2", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", .init = sdp3430_twl4030_voice_init, .ops = &sdp3430_voice_ops, }, }; This now means we now no longer have to export all the CODEC | DAI | platform structure symbols around and can keep them static. We also manage to delete a few headers as a result of this change. Since we are now using dev_name or DAI name (for single devices with multiple DAIs) I've taken the opportunity to standardise the ASoC component naming. It's currently very ad-hoc and platform specific, so I've now added some useful prefixes to help. e.g. So we now have :- "-codec" prefix for codecs, e.g. "wm8731-codec" "-dai" prefix for CPU DAI's e.g. "pxa-i2s-dai" "-hifi", -"voice", etc for CODEC DAIs e.g. "wm8731-hifi" "-pcm-audio" for platform DMA, e.g. "pxa-pcm-audio" for PXA DMA (although most platform DMA controllers used this anyway). This naming standard will also come in handy for device tree too. The only other change I've made is fixing the twl4030 and DAC33 codec probe and remove (as spotted by Peter). I've also received and applied numerous patches from Jarkko, Timur, Ryan, Chanwoo, Joonyoug, Kyungmin, Jassi and Seungwhan. The multi-component code can be found here :- git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git topic/multi-component As usual, please give this some testing (especially ASoC platform maintainers) and if all is OK we can upstream some time next week (or the week after). Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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Re: [alsa-devel] Which project to choose?
by Raymond Yau 22 Jul '10

22 Jul '10

2010/7/22 Chris <cpollock(a)embarqmail.com> > On Thu, 2010-07-22 at 11:03 +0800, Raymond Yau wrote: > > > > > > 2010/7/22 Chris <cpollock(a)embarqmail.com> > > > > > > > > > > AFAIK, the most buggy PA client is pavucontrol because > > > > > > 1) it has those vumeters which need to get audio data from > > the > > > playback and capture devices of the sound card , > > > 2) when you move the slider to left ( over 100%) and this > > force PA to > > > provide software gain > > > > > > You should notice that the new gnome-volume-control is > > actually a > > > simplified version of pavucontrol > > > > > > Hmm, I just pulled up PulseAudio Volume Control and see no > > vumeters, I > > can't figure out though why the drive thrashing at various > > times whether > > I have a stream playing or not. > > > > > > Did you click the input device tab page of the gnome volume control ? > > > > The vumeter is "Input level" above the volume control slider ? > > > > You should noticed that the following message appear in the pulseaudio > > log > > > > I: source-output.c : Created output 0 "Peak detect" on alsa_input > > > > I.source-output:c application.id = "org.gnome.volume.control" > > I.source-output:c media.name = "Peak detect" > > I.source-output:c application.name = "GNOME Volume control dialog" > > > > when you using mplayer to play music and open pavucontrol to > > > > pavucontrol sometime abort suddenly when you aborte mplayer but this > > is not 100% reproducible > > > > pavucontrol > > ** ( pavucontrol:3232) : DEBUG: failed to initialize device manager > > extension: No such extension > > ** (pavucontrol:3056) : DEBUG: Ignoring sink-input due to it being > > designated as an event and thuse handled by the event widget > > > > > > The point is after the complete intergration of pulseaudio in to the > > system ( libcanberra, mixer-applet , ... ) > > > > I think "pulseaudio -k; pulseaudio -vvvv" is no longer the correct way > > to debug since there are lots of PA clients running at the desktop > > > > You have to ask the mandriva maintainer how to increase the debug > > level of PA server and have to perform a logout/login to debug the PA > > server > > Thanks Raymond, so I should contact Colin and go from there? I'm not > sure what to even tell him though. I thank you for all your help. > > Chris > There are over 15 PA clients running when you are using gnome desktop running "pulseaudio -k " to shutdown PA server , this kill all the PA clients restart PA server by "pulseaudio -vvvv" , only one or two PA clients (e.g. consolekit, .. ) can connected back to PA server This command "pulseaudio -k;pulseaudio -vvvv" cannot provide a normal environment to debug PA server since most of PA clients are killed

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[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix compile error when registering a DAI
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify some code which register a DAI with ASoC core according to modified registration API of DAI. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 3 +-- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index 69c41eb..007ed3a 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -268,8 +268,7 @@ static int __init aquila_init(void) return -ENOMEM; /* register voice DAI here */ - ret = snd_soc_register_dai(&aquila_snd_device->dev, - 0, &voice_dai); + ret = snd_soc_register_dai(&aquila_snd_device->dev, &voice_dai); if (ret) return ret; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index f4fbc0e..b3866d5 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -750,7 +750,7 @@ int s3c_i2sv2_register_dai(struct device *dev, int id, drv->suspend = s3c2412_i2s_suspend; drv->resume = s3c2412_i2s_resume; - return snd_soc_register_dai(dev, id, drv); + return snd_soc_register_dai(dev, drv); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); -- 1.6.3.3

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[alsa-devel] multi-component: determine which DAI is active?
by Timur Tabi 22 Jul '10

22 Jul '10

Liam, In order to get fsl_dma.c to work with the WM8776, which has separate DAIs for playback and capture, I've had to do this: if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't allocate playback dma buffer\n"); return ret; } } if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't allocate capture dma buffer\n"); return ret; } } That is, I need to check if the 'substream' pointer has been initialized. This looks ugly to me. In an earlier version of ASoC, I would have done this: if (dai->playback.channels_min) { ... if (dai->capture.channels_min) { ... But dai->playback and dai->capture no longer exist. How should I handle this?

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[alsa-devel] [PATCH] ASoC: multi-component - Add Goni sound driver
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch add sound support for the Goni board based on S5PV210. The Goni board is based on Samsung SoC(S5PV210) and include WM8994 codec over I2S to support sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Goni board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Goni board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 ++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/goni_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/goni_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index c09b402..1cdc37b 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -140,3 +140,12 @@ config SND_S5PC110_SOC_AQUILA_WM8994 help Say Y if you want to add support for SoC audio on aquila with the WM8994. + +config SND_S5PV210_SOC_GONI_WM8994 + tristate "SoC I2S Audio support for GONI - WM8994" + depends on SND_S3C24XX_SOC && MACH_GONI + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on goni + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 02dd12c..47ed6d7 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -31,6 +31,7 @@ snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-aquila-wm8994-objs := aquila_wm8994.o +snd-soc-goni-wm8994-objs := goni_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -45,3 +46,4 @@ obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o +obj-$(CONFIG_SND_S5PV210_SOC_GONI_WM8994) += snd-soc-goni-wm8994.o diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c new file mode 100644 index 0000000..20bcaab --- /dev/null +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -0,0 +1,298 @@ +/* + * goni_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +static struct snd_soc_card goni; +static struct platform_device *goni_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Left Spk", NULL), + SND_SOC_DAPM_SPK("Ext Right Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route goni_dapm_routes[] = { + {"Ext Left Spk", NULL, "SPKOUTLP"}, + {"Ext Left Spk", NULL, "SPKOUTLN"}, + + {"Ext Right Spk", NULL, "SPKOUTRP"}, + {"Ext Right Spk", NULL, "SPKOUTRN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add goni specific widgets */ + snd_soc_dapm_new_controls(codec, goni_dapm_widgets, + ARRAY_SIZE(goni_dapm_widgets)); + + /* set up goni specific audio routes */ + snd_soc_dapm_add_routes(codec, goni_dapm_routes, + ARRAY_SIZE(goni_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&goni, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int goni_hifi_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; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops goni_hifi_ops = { + .hw_params = goni_hifi_hw_params, +}; + +static int goni_voice_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; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "goni-voice-dai", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops goni_voice_ops = { + .hw_params = goni_voice_hw_params, +}; + +static struct snd_soc_dai_link goni_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "s3c64xx-i2s-v4-dai", + .codec_dai_name = "wm8994-hifi", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .init = goni_wm8994_init, + .ops = &goni_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_name = "goni-voice-dai", + .codec_dai_name = "wm8994-voice", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .ops = &goni_voice_ops, +}, +}; + +static struct snd_soc_card goni = { + .name = "goni", + .dai_link = goni_dai, + .num_links = ARRAY_SIZE(goni_dai), +}; + +static int __init goni_init(void) +{ + int ret; + + if (!machine_is_goni()) + return -ENODEV; + + goni_snd_device = platform_device_alloc("soc-audio", -1); + if (!goni_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(goni_snd_device, &goni); + ret = platform_device_add(goni_snd_device); + + if (ret) + platform_device_put(goni_snd_device); + + return ret; +} + +static void __exit goni_exit(void) +{ + platform_device_unregister(goni_snd_device); +} + +module_init(goni_init); +module_exit(goni_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify the wrong field name on Aquila board. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index f6f0e60..69c41eb 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -242,7 +242,7 @@ static struct snd_soc_dai_link aquila_dai[] = { }, { .name = "WM8994 Voice", .stream_name = "Voice", - .cpu_dai_drv = "aquila-voice-dai", + .cpu_dai_name = "aquila-voice-dai", .codec_dai_name = "wm8994-voice", .platform_name = "s3c24xx-pcm-audio", .codec_name = "wm8994-codec", -- 1.6.3.3

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[alsa-devel] problems about converting sample rates when using ALSA-LIB
by longer 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-lib-1.0.23), I found that : It mmap a address from my driver , but writing audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data . But I don't know why it don't write audio data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

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[alsa-devel] problems about converting sample rate when using ALSA lib
by longer zhou (周龙) 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-driver-1.0.23), I found that : It mmap a address from my driver , but writting audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data .But I don't know why it don't write audo data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

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[alsa-devel] add support for Codec: Conexant ID 5068 (not supported by ALSA yet)
by ccamacho＠linuxmail.org 22 Jul '10

22 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

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[alsa-devel] Small memory leak in snd_pcm_ladspa_init()
by John Lindgren 22 Jul '10

22 Jul '10

Hello, Valgrind shows up a small memory leak in libasound when using a LADSPA plugin. Sorry, I don't have a patch this time. ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,976 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,977 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ... ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,980 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,981 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ... and so on. -- John

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