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

[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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[alsa-devel] [PATCH] ALSA: AT73C213: Rectify misleading comments.
by Peter Rosin 15 Nov '10

15 Nov '10

The Atmel SSC can divide by even numbers, not only powers of two and the given values are binary, not hexadecimal. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/spi/at73c213.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) I'm working on a driver based on AT73C213 and have had to look up if the below nits were code problems or comment problems. The Atmel specs tells me that the comments are wrong and this patch fixes those comments. Cheers and thanks, Peter diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 1bc56b2..548e17a 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -155,7 +155,7 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) if (max_tries < 1) max_tries = 1; - /* ssc_div must be a power of 2. */ + /* ssc_div must be even. */ ssc_div = (ssc_div + 1) & ~1UL; if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) { @@ -621,7 +621,7 @@ static int snd_at73c213_line_capture_volume_info( { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; - /* When inverted will give values 0x10001 => 0. */ + /* When inverted will give values 10001b => 0. */ uinfo->value.integer.min = 14; uinfo->value.integer.max = 31; @@ -634,7 +634,7 @@ static int snd_at73c213_aux_capture_volume_info( { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; - /* When inverted will give values 0x10001 => 0. */ + /* When inverted will give values 10001b => 0. */ uinfo->value.integer.min = 14; uinfo->value.integer.max = 31; -- 1.7.2.3

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[alsa-devel] [PATCHv2] ASoC: Add optional name_prefix for codec kcontrol, widget and route names
by Jarkko Nikula 15 Nov '10

15 Nov '10

There is a need to prefix codec kcontrol, widget and internal route names in an ASoC machine that has multiple codecs with conflicting names. The name collision would occur when codec drivers try to registering kcontrols with the same name or when building audio paths. This patch introduces optional prefix_map into struct snd_soc_card. With it machine drivers can specify a unique name prefix to each codec that have conflicting names with anothers. Prefix to codec is matched with codec name. Following example illustrates a machine that has two same codec instances. Name collision from kcontrol registration is avoided by specifying a name prefix "foo" for the second codec. As the codec widget names are prefixed then second audio map for that codec shows a prefixed widget name. static const struct snd_soc_dapm_route map0[] = { {"Spk", NULL, "MONO"}, }; static const struct snd_soc_dapm_route map1[] = { {"Vibra", NULL, "foo MONO"}, }; static struct snd_soc_prefix_map codec_prefix[] = { { .dev_name = "codec.2", .name_prefix = "foo", }, }; static struct snd_soc_card card = { ... .prefix_map = codec_prefix, .num_prefixes = ARRAY_SIZE(codec_prefix), }; Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- This goes on top of "ASoC: Extend DAPM to handle cross-device paths" set: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033583.h… Basically updated prefix_map to snd_soc_card idea in RFC_iv/iv: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-October/033240.ht… Now prefix is not tied to snd_soc_dai_link and registration of dailess codecs will come after this. Otherwise the same but codec_name in snd_soc_prefix_map changed to dev_name. --- include/sound/soc.h | 13 +++++++++++++ sound/soc/soc-core.c | 35 +++++++++++++++++++++++++++++++++-- sound/soc/soc-dapm.c | 34 ++++++++++++++++++++++++++++++++-- 3 files changed, 78 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8470059..346c59e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -450,6 +450,7 @@ struct snd_soc_cache_ops { /* SoC Audio Codec device */ struct snd_soc_codec { const char *name; + const char *name_prefix; int id; struct device *dev; struct snd_soc_codec_driver *driver; @@ -577,6 +578,11 @@ struct snd_soc_dai_link { struct snd_soc_ops *ops; }; +struct snd_soc_prefix_map { + const char *dev_name; + const char *name_prefix; +}; + /* SoC card */ struct snd_soc_card { const char *name; @@ -611,6 +617,13 @@ struct snd_soc_card { struct snd_soc_pcm_runtime *rtd; int num_rtd; + /* + * optional map of kcontrol, widget and path name prefixes that are + * associated per device + */ + struct snd_soc_prefix_map *prefix_map; + int num_prefixes; + struct work_struct deferred_resume_work; /* lists of probed devices belonging to this card */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a16096..9c0b2c7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1397,6 +1397,23 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } } +static void soc_set_name_prefix(struct snd_soc_card *card, + struct snd_soc_codec *codec) +{ + int i; + + if (card->prefix_map == NULL) + return; + + for (i = 0; i < card->num_prefixes; i++) { + struct snd_soc_prefix_map *map = &card->prefix_map[i]; + if (map->dev_name && !strcmp(codec->name, map->dev_name)) { + codec->name_prefix = map->name_prefix; + break; + } + } +} + static void rtd_release(struct device *dev) {} static int soc_probe_dai_link(struct snd_soc_card *card, int num) @@ -1406,6 +1423,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + const char *temp; int ret; dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); @@ -1440,6 +1458,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the CODEC */ if (!codec->probed) { codec->dapm.card = card; + soc_set_name_prefix(card, codec); if (codec->driver->probe) { ret = codec->driver->probe(codec); if (ret < 0) { @@ -1493,11 +1512,15 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* now that all clients have probed, initialise the DAI link */ if (dai_link->init) { + /* machine controls, routes and widgets are not prefixed */ + temp = rtd->codec->name_prefix; + rtd->codec->name_prefix = NULL; ret = dai_link->init(rtd); if (ret < 0) { printk(KERN_ERR "asoc: failed to init %s\n", dai_link->stream_name); return ret; } + rtd->codec->name_prefix = temp; } /* Make sure all DAPM widgets are instantiated */ @@ -2076,14 +2099,22 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; + char prefixed_name[44], *name; int err, i; for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); + if (codec->name_prefix) { + snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", + codec->name_prefix, control->name); + name = prefixed_name; + } else { + name = control->name; + } + err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, control->name, err); + codec->name, name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ec43f9..27e3f41 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1336,6 +1336,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) kfree(p->long_name); kfree(p); } + kfree(w->name); kfree(w); } } @@ -1384,11 +1385,25 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; - const char *sink = route->sink; + const char *sink; const char *control = route->control; - const char *source = route->source; + const char *source; + char prefixed_sink[80]; + char prefixed_source[80]; int ret = 0; + if (dapm->codec->name_prefix) { + snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", + dapm->codec->name_prefix, route->sink); + sink = prefixed_sink; + snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", + dapm->codec->name_prefix, route->source); + source = prefixed_source; + } else { + sink = route->sink; + source = route->source; + } + /* * find src and dest widgets over all widgets but favor a widget from * current DAPM context @@ -2027,10 +2042,25 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; + size_t name_len; if ((w = dapm_cnew_widget(widget)) == NULL) return -ENOMEM; + name_len = strlen(widget->name) + 1; + if (dapm->codec->name_prefix) + name_len += 1 + strlen(dapm->codec->name_prefix); + w->name = kmalloc(name_len, GFP_KERNEL); + if (w->name == NULL) { + kfree(w); + return -ENOMEM; + } + if (dapm->codec->name_prefix) + snprintf(w->name, name_len, "%s %s", + dapm->codec->name_prefix, widget->name); + else + snprintf(w->name, name_len, "%s", widget->name); + dapm->n_widgets++; w->dapm = dapm; w->codec = dapm->codec; -- 1.7.2.3

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[alsa-devel] ASoC: WM8804: fsi-wm8804
by Grzegorz Daniluk 15 Nov '10

15 Nov '10

Hi, I'm trying to use the wm8804 codec in my SH4-based system (Renesas SH7724 to be precise). I have used the latest kernel 2.6.37-rc1 where there is the codec driver for wm8804. However, I have to write my own fsi-wm8804.c (like e.g. fsi-da7210.c). My code attached below. After compiling it with the rest of the kernel sources or compiling as an external module my alsa still does not see any soundcards. However, by running i2cdetect I can see that the device is correctly taken by the codec driver so the problem is somewhere with my fsi-wm8804 probably. Could you take a look at this ? Did I miss something from new ASoC api so that this module could correctly talk with wm8804 codec driver ? Also (when compiled as modules) after loading and doing lsmod I could see that the wm8804 module is not used by fsi-wm8804 module (as it was when I was playing with wm8940 on previous kernel versions). best regards, Greg #include <linux/platform_device.h> #include <sound/sh_fsi.h> static int fsi_wm8804_init(struct snd_soc_codec *codec) { return 0; } static int fsi_wm8804_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 *dai = rtd->codec_dai; printk("%s\n", __FUNCTION__); if(substream->stream == SNDRV_PCM_STREAM_CAPTURE) { snd_soc_dai_set_pll(dai, 0, 0, 12000000, 94310400); } else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { snd_soc_dai_set_pll(dai, 0, 0, 12000000, 98304000); } snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); return 0; } static struct snd_soc_ops fsi_wm8804_ops = { .hw_params = fsi_wm8804_hw_params, }; static struct snd_soc_dai_link fsi_wm8804_dai = { .name = "WM8804", .stream_name = "WM8804", .cpu_dai_name = "fsib-dai", /* FSI B */ .codec_dai_name = "wm8804-spdif", .platform_name = "sh_fsi.0", .codec_name = "wm8804.0-003a", /*0x3a is codec I2C addr*/ .init = fsi_wm8804_init, .ops = &fsi_wm8804_ops, }; static struct snd_soc_card fsi_soc_card = { .name = "FSI (WM8804)", .dai_link = &fsi_wm8804_dai, .num_links = 1, }; static struct platform_device *fsi_wm8804_snd_device; static int __init fsi_wm8804_sound_init(void) { int ret; printk("%s\n", __FUNCTION__); fsi_wm8804_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); if (!fsi_wm8804_snd_device) return -ENOMEM; platform_set_drvdata(fsi_wm8804_snd_device, &fsi_soc_card); ret = platform_device_add(fsi_wm8804_snd_device); if (ret) platform_device_put(fsi_wm8804_snd_device); return ret; } static void __exit fsi_wm8804_sound_exit(void) { printk("%s\n", __FUNCTION__); platform_device_unregister(fsi_wm8804_snd_device); } module_init(fsi_wm8804_sound_init); module_exit(fsi_wm8804_sound_exit); /* Module information */ MODULE_DESCRIPTION("ALSA SoC FSI WM8804"); MODULE_LICENSE("GPL")

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[alsa-devel] [PATCH 2.6.37 1/2] ASoC: Fix dapm_seq_compare() for multi-component
by Mark Brown 15 Nov '10

15 Nov '10

Ensure that we keep all widget powerups in DAPM sequence by making the CODEC the last thing we compare on rather than the first thing. Also fix the fact that we're currently comparing the widget pointers rather than the CODEC pointers when we do the substraction so we won't get stable results. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8352430..bc2ec06 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -706,12 +706,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, int sort[]) { - if (a->codec != b->codec) - return (unsigned long)a - (unsigned long)b; if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; if (a->reg != b->reg) return a->reg - b->reg; + if (a->codec != b->codec) + return (unsigned long)a->codec - (unsigned long)b->codec; return 0; } -- 1.7.1

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[alsa-devel] [patch 1/1] Kirkwood/ASoC: Add support for OpenRD Ultimate
by Arnaud Patard 15 Nov '10

15 Nov '10

OpenRD Ultimate & Client are similar machines so enable OpenRD client sound support on Ultimate too Tested-by: Robas Teodor <teodor.robas(a)gmail.com> Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: sound-2.6/arch/arm/mach-kirkwood/openrd-setup.c =================================================================== --- sound-2.6.orig/arch/arm/mach-kirkwood/openrd-setup.c 2010-11-12 17:58:41.000000000 +0100 +++ sound-2.6/arch/arm/mach-kirkwood/openrd-setup.c 2010-11-12 17:59:06.000000000 +0100 @@ -94,7 +94,7 @@ kirkwood_i2c_init(); - if (machine_is_openrd_client()) { + if (machine_is_openrd_client() || machine_is_openrd_ultimate()) { i2c_register_board_info(0, i2c_board_info, ARRAY_SIZE(i2c_board_info)); kirkwood_audio_init(); Index: sound-2.6/sound/soc/kirkwood/kirkwood-openrd.c =================================================================== --- sound-2.6.orig/sound/soc/kirkwood/kirkwood-openrd.c 2010-11-12 17:59:17.000000000 +0100 +++ sound-2.6/sound/soc/kirkwood/kirkwood-openrd.c 2010-11-12 17:59:31.000000000 +0100 @@ -86,7 +86,7 @@ { int ret; - if (!machine_is_openrd_client()) + if (!machine_is_openrd_client() && !machine_is_openrd_ultimate()) return 0; openrd_client_snd_device = platform_device_alloc("soc-audio", -1); Index: sound-2.6/sound/soc/kirkwood/Kconfig =================================================================== --- sound-2.6.orig/sound/soc/kirkwood/Kconfig 2010-11-12 17:59:51.000000000 +0100 +++ sound-2.6/sound/soc/kirkwood/Kconfig 2010-11-12 18:00:27.000000000 +0100 @@ -11,7 +11,7 @@ config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && MACH_OPENRD_CLIENT + depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help

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[alsa-devel] [PATCH] ASoC: Prevent system suspend while debouncing wakeup capable GPIO jacks
by Mark Brown 15 Nov '10

15 Nov '10

If the device associated with a GPIO jack is wakeup capable then disable suspend while we're debouncing the jack so that we skip suspends that race with the jack. Note that currently the GPIO based jack has a CODEC associated with it which we're using right now. These jacks should be reparented against the card itself and this code adjusted. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4d95abb..619061d 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -209,6 +209,10 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; + struct device *dev = gpio->jack->codec->card->dev; + + if (device_may_wakeup(dev)) + pm_wakeup_event(dev, gpio->debounce_time + 50); schedule_delayed_work(&gpio->work, msecs_to_jiffies(gpio->debounce_time)); -- 1.7.2.3

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[alsa-devel] snd-soc - platform_device_add failed
by Vaclav Peroutka 13 Nov '10

13 Nov '10

Hello, I am trying to create alsa driver for MAX9880 which is connected to S3C6410. I "copied" the initialization code from other examples in sound/soc/ but during initialization I am getting "platform_device_add failed". I attached the code for kernel 2.6.29, ALSA 1.0.18a. For me it does not have anything which can fail. But I never wrote drivers for ALSA. I have another question. In the kernel code sound/soc/s3c_64xx/, there is a plenty of s3c_i2s files. Different versions but no description, which version should I use. Does anybody know, which one is the right version ? Thank you in advance, Vaclav

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[alsa-devel] ALSA-SOC support for OpenRD Client/Ultimate
by Robas, Teodor 13 Nov '10

13 Nov '10

Hello everyone, I am writing to this list in the hope that somebody can clear me out if this platform is supported and I am doing something wrong, or if it is still work in progress. The drivers that I think should work with my hardware (OpenRD ultimate) are located in sound/soc/kirkwood/*. Sound on this device worked (partially) with the default software it came but not with the latest about 10 kernels I have tried. I added bellow the output of alsa-info.sh script. If needed I can look for more or try to fix it (with some help). Best Regards, Teodor !!################################ !!ALSA Information Script v 0.4.59 !!################################ !!Script ran on: Thu Nov 11 06:27:39 UTC 2010 !!Linux Distribution !!------------------ Debian GNU/Linux squeeze/sid \n \l !!DMI Information !!--------------- Manufacturer: Product Name: !!Kernel Information !!------------------ Kernel release: 2.6.37-rc1 Operating System: GNU/Linux Architecture: armv5tel Processor: unknown SMP Enabled: No !!ALSA Version !!------------ Driver version: 1.0.23 Library version: 1.0.23 Utilities version: 1.0.23 !!Loaded ALSA modules !!------------------- !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- --- no soundcards --- !!PCI Soundcards installed in the system !!-------------------------------------- !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 33 Nov 11 08:23 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY aplay: device_list:235: no soundcards found... ARECORD arecord: device_list:235: no soundcards found... !!Amixer output !!------------- !!Alsactl output !!------------- --startcollapse-- --endcollapse-- !!All Loaded Modules !!------------------ Module snd_soc_kirkwood snd_soc_core snd_pcm snd_timer snd snd_page_alloc bridge stp llc nfsd lockd nfs_acl auth_rpcgss sunrpc exportfs loop xgifb fb cfbcopyarea mv_cesa cfbimgblt aes_generic cfbfillrect sd_mod crc_t10dif usb_storage sata_mv libata ehci_hcd scsi_mod usbcore mv643xx_eth libphy nls_base !!ALSA/HDA dmesg !!------------------ PS: please note that none of the snd* modules are loaded by default. PPS: Somewhere in the code (sound/soc/kirkwood/kirkwood-openrd.c) there is a check: machine_is_openrd_client() but in my case it should be machine_is_openrd_ultimate(). I removed this check anyway. -- _ ( ) ascii ribbon campaign - against html e-mail X www.asciiribbon.org - against proprietary attachments / \

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[alsa-devel] wrong channel mappings for HDMI audio
by Dan Christensen 13 Nov '10

13 Nov '10

[System specs at end.] When I try to use 5.1 audio over HDMI, the front left and front right channels are correct, but the others are wrong: - surround (rear) left and center are swapped - surround (rear) right and LFE are swapped This has been tested using "speaker-test -c6 -Dhw:0,3 -twav" and also by playing http://www.avenard.org/files/media/videotest/audiotest/idch_AviaSurroundTes… http://www.avenard.org/files/media/videotest/audiotest/idch_AviaSurroundTes… using mplayer and mythtv. They play correctly with digital passthrough: AAC and DTS for mythtv, and DTS for mplayer. Jean-Yves Avenard reports that they play correctly using 5.1 analog output on the same motherboard, but I haven't tested that myself. Let me know if this would be helpful, e.g. if it is hard to believe that analog has the correct mapping while HDMI doesn't. James Courtier-Dutton suggested that I report this here, in the hopes that ALSA can be taught the correct mapping for my sound card. I'm also interested in the best way to work around this in the meantime. Thanks for any help! Dan System specs: Motherboard: Gigabyte GA-E7AUM-DS2H http://gigabyte.com/products/product-page.aspx?pid=2946#ov Up-to-date Debian squeeze with 2.6.32-5-686-bigmem kernel. ALSA packages installed: alsa-base 1.0.23+dfsg-2 ALSA driver configuration files alsa-oss 1.0.17-4 ALSA wrapper for OSS applications alsa-source 1.0.23+dfsg-2 ALSA driver sources alsa-utils 1.0.23-3 Utilities for configuring and using ALSA alsaplayer-alsa 0.99.80-5+b1 PCM player designed for ALSA (ALSA output module) alsaplayer-common 0.99.80-5+b1 PCM player designed for ALSA (common files) alsaplayer-oss 0.99.80-5+b1 PCM player designed for ALSA (OSS output module) alsaplayer-text 0.99.80-5+b1 PCM player designed for ALSA (text version) libasound2 1.0.23-2.1 shared library for ALSA applications linux-sound-base 1.0.23+dfsg-2 base package for ALSA and OSS sound systems xmms2-plugin-alsa 0.7DrNo+dfsg-2 XMMS2 - ALSA output Pioneer VSX-820-K receiver, which has 5 speakers connected (no subwoofer), receiving input over HDMI. asound.conf: pcm.!default { type hw card 0 device 3 } lspci -vv output: 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) Subsystem: Giga-byte Technology Device a102 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 21 Region 0: Memory at d9180000 (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 Kernel modules: snd-hda-intel [To save you scrolling: there's nothing else after the /proc output.] Output of cat /proc/asound/card0/codec#3 [below is #0 in case it helps]: Codec: Nvidia MCP7A HDMI Address: 3 Function Id: 0x1 Vendor Id: 0x10de0007 Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x18560110: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560121: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560122: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560123: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560124: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Output of cat /proc/asound/card0/codec#0: Codec: Realtek ALC889A Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0885 Subsystem Id: 0x1458a002 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled GenLevel Digital category: 0x2 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x1c 0x1c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c40: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181344f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4005e601: [N/A] Line Out at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x01cb7160: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Processing Coefficient: 0xc128 Coefficient Index: 0x02 Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=32, direct=0, val=0 Unsolicited: tag=00, enabled=0 Connection: 0 Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x25 0x0b

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