July 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Cirrus Logic CS4270 support coming soon
by Timur Tabi 18 Jul '07

18 Jul '07

Hi Everyone, I just wanted to let everyone know that I will soon be submitting an ASoC driver for the Cirrus Logic CS4270 codec. I have the driver working in-house on an older version of ALSA. As soon as I finish work on my current project, I will clean up the driver, update to the latest code, and submit it. If anyone is using this codec on their hardware, please let me know. The CS4270 supports many different configurations, but my board only enables some of them.

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[alsa-devel] substream->private_data == NULL in hw_params
by Florian Demski 17 Jul '07

17 Jul '07

Hi there, after many reboots I finally fixed an error in a driver but I am not sure if there ain't an underlying problem which causes this error. The initialization is done with struct snd_pcm *pcm; int err; if((err = snd_pcm_new(chip->pod->line6.card, (char *)chip->pod->line6.devname, 0, 1, 1, &pcm)) < 0) return err; pcm->private_data = chip; pcm->private_free = pod_cleanup_pcm; chip->pcm = pcm; However in hw_param, substream->private_data is NULL and I have to get the right value with this construct: static int snd_pod_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { int ret; struct snd_pod_pcm *chip = snd_pcm_substream_chip(substream); /* -- Florian Demski [FD] */ /* don't ask me why, but this fixes the bug on my machine */ if ( chip == NULL ) { if ( substream->pcm == NULL ) return -ENOMEM; if ( substream->pcm->private_data == NULL ) return -ENOMEM; substream->private_data = substream->pcm->private_data; chip = snd_pcm_substream_chip( substream ); } /* -- [FD] end */ Somehow only my machine produces this error, I tried switching off SMP and PREEMPT, but with no other result. Or is this possibly an (undocumented) feature in alsa? I'd be happy if actually some of you experienced guys might take a look at that driver: SVN https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/trunk Browse https://line6linux.svn.sourceforge.net/viewvc/line6linux/driver/trunk/ Thanks in advance, Florian

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[alsa-devel] [PATCH] hda: Enable SPDIF in/out on some stac9205 boards
by Matthew Ranostay 17 Jul '07

17 Jul '07

Added support for some boards with SPDIF in/out, and cleaned up the GPIO enable function. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 631a50e624df pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jul 16 11:01:30 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Mon Jul 16 16:34:46 2007 -0400 @@ -44,6 +44,7 @@ enum { enum { STAC_9205_REF, + STAC_M43xx, STAC_9205_MODELS }; @@ -218,7 +219,6 @@ static hda_nid_t stac9205_pin_nids[12] = 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x16, 0x17, 0x18, 0x21, 0x22, - }; static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, @@ -773,7 +773,8 @@ static unsigned int ref9205_pin_configs[ }; static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { - ref9205_pin_configs, + [STAC_REF] = ref9205_pin_configs, + [STAC_M43xx] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -784,6 +785,10 @@ static struct snd_pci_quirk stac9205_cfg /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8, + "Dell Precision", STAC_M43xx), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff, + "Dell Precision", STAC_M43xx), {} /* terminator */ }; @@ -813,48 +818,52 @@ static int stac92xx_save_bios_config_reg return 0; } +static void stac92xx_set_config_reg(struct hda_codec *codec, hda_nid_t pin_nid, unsigned int pin_config) +{ + int i; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + i = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, + 0x00); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", pin_nid, i); +} + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; - unsigned int pin_cfg; - - if (! spec->pin_nids || ! spec->pin_configs) - return; - - for (i = 0; i < spec->num_pins; i++) { - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - spec->pin_configs[i] & 0x000000ff); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (spec->pin_configs[i] & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (spec->pin_configs[i] & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - spec->pin_configs[i] >> 24); - pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); - } -} - -static void stac92xx_enable_eapd(struct hda_codec *codec) -{ - /* Configure GPIO0 as output */ + if (!spec->pin_configs) + return; + + for (i=0; i < spec->num_pins; i++) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); +} + +static void stac92xx_enable_gpio_mask(struct hda_codec *codec,int gpio_mask,int gpio_data) +{ + /* Configure GPIOx as output */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ + AC_VERB_SET_GPIO_DIRECTION, gpio_mask); + /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ + /* Assert GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ + AC_VERB_SET_GPIO_DATA, gpio_data); + /* Enable GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + AC_VERB_SET_GPIO_MASK, gpio_mask); } /* @@ -2233,7 +2242,7 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); + stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001); /* GPIO0 High = Enable EAPD */ err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { @@ -2258,7 +2267,7 @@ static int patch_stac9205(struct hda_cod static int patch_stac9205(struct hda_codec *codec) { struct sigmatel_spec *spec; - int err; + int err, gpio_mask, gpio_data; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2295,8 +2304,19 @@ static int patch_stac9205(struct hda_cod spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); - + + if (spec->board_config == STAC_M43xx) { + /* Enable SPDIF in/out */ + stac92xx_set_config_reg(codec, 0x1f, 0x01441030); + stac92xx_set_config_reg(codec, 0x20, 0x1c410030); + + gpio_mask = 0x00000007; /* GPIO0-2 */ + gpio_data = 0x00000005; /* GPIO0 High = EAPD, GPIO1 Low = DRM, GPIO2 High = Headphone Mute */ + } + else + gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + + stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { if (spec->board_config < 0) {

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[alsa-devel] Input/output error - Confusion regarding period size and interrupts
by Pharaoh . 16 Jul '07

16 Jul '07

Hi The hardware I am working on expects 8K bytes to be DMAed before giving an interrupt. Only after a block of 8K bytes is DMAed, I get an end of the block interrupt and I enqueue next 8K bytes for processing. The params look like these: static snd_pcm_hardware_t snd_omap_alsa_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_MMAP), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 ), .rate_min = 44100, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 8192*8, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, .periods_min = 2, //ping pong buffer .periods_max = 128, .fifo_size = 0, }; I just copied it from another driver ;-). But now I am getting an input/output error after first block of data i.e. 8K bytes are DMAed. Does it have anything to do with the above struct? I have discussed this earlier too, but now attacking it once again after long gap. -pharaoh.

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[alsa-devel] [Alsa-devel] xrun problem
by Gao, Jonathan 16 Jul '07

16 Jul '07

Hi, I am writing a PCM driver on a Soc for 2.6.10 kernel. The test code I'm using is from alsa-lib-1.0.6rc1/test/pcm.c. The runtime log gives me like this... ----------------------------------------------------------------- open callback: hw params callback: alloc result = 1, buffer bytes = 4608 /* --------- HW params ------------- */ format = 2, rate = 44100, channels= 2, period_size = 288, periods = 4, buffer_size = 1152 prepare callback /* --------- Runtime Info ---------- */ state = 1 (SNDRV_PCM_STATE_SETUP), hw_ptr = 0, appl_ptr = 0, avail_min = 288 avail_max = 0, hw_ptr_base = 0, hw_ptr_interrupt = 0 trigger callback: START Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 0, appl_ptr = 1152, avail_min = 288 avail_max = 0, hw_ptr_base = 0, hw_ptr_interrupt = 0 pointer callback: hw pointer: 288 (return value of pointer callback) Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 288, appl_ptr = 1152, avail_min = 288 avail_max = 288, hw_ptr_base = 0, hw_ptr_interrupt = 288 pointer callback: hw pointer: 576 (return value of pointer callback) Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 576, appl_ptr = 1152, avail_min = 288 avail_max = 576, hw_ptr_base = 0, hw_ptr_interrupt = 576 pointer callback: hw pointer: 864 (return value of pointer callback) Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 864, appl_ptr = 1152, avail_min = 288 avail_max = 864, hw_ptr_base = 0, hw_ptr_interrupt = 864 pointer callback: hw pointer: 0 (return value of pointer callback) trigger callback: STOP prepare callback /* --------- Runtime Info ---------- */ state = 4 (SNDRV_PCM_STATE_XRUN), hw_ptr = 1152, appl_ptr = 1152, avail_min = 288 avail_max = 1152, hw_ptr_base = 1152, hw_ptr_interrupt = 1152 ----------------------------------------------------------------- The kernel hangs after prepare callback, from strace, I can see a endless ioctl(4, ..., ...) repeated. Does anyone have a few pointers on what is going wrong? Best Regards, Jonathan

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Re: [alsa-devel] [PATCH] Fix request_irq error handling
by Ash Willis 16 Jul '07

16 Jul '07

Hah, um, ignore me. It seems that my c needs some work ;) In if(some_expression), I always though it is true if the expression is > 0, not != 0. If anything, that was a CodingStyle patch because it lets us see exactly what we're getting from request_irq but I can see that it is totally unnecessary :) > ----- Original Message ----- > From: "Ash Willis" <ashwillis(a)programmer.net> > To: alsa-devel(a)alsa-project.org > Subject: [alsa-devel] [PATCH] Fix request_irq error handling > Date: Sat, 14 Jul 2007 02:29:19 -0500 > > > request_irq returns 0 on success and a negative code on failure. > Many ALSA drivers incorrectly check for failure by testing for a > positive code instead of a negative one. > > Signed-Off-By: Ash Willis <ashwillis(a)programmer.net> > > > -- > We've Got Your Name at http://www.mail.com! > Get a FREE E-mail Account Today - Choose From 100+ Domains > << request_irq_error_handling.patch >> > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains

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Re: [alsa-devel] No sound on Medion laptop (intel 82801G)
by albox＠free.fr 16 Jul '07

16 Jul '07

I have added a note for this bug on : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3160 I'm not really sure but it seems that the file /etc/modprobe.d/alsa-base is not loaded. http://ubuntuforums.org/showthread.php?p=3026300#post3026300 Any idea ? Thanks

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[alsa-devel] Install of alsa 1.0.14.final on Fedora 7
by stan 15 Jul '07

15 Jul '07

On Fri, 13 Jul 2007 08:53:47 -0700 tdavis(a)dsl-only.net wrote: > For RPM based systems, I have a build environment for the drivers, > libs, and utils packages at > http://members.dsl-only.net/~tdavis/my-build.tar.bz2. You need to > be root to build them, but they won't interfere with your rpm > distribution. THe driver rpm will also make a backup of your > current drivers prior to installation. > > Tobin > Hi Tobin, I unpacked the archive and typed make. It created 4 RPMs. Two of them installed, alsa-lib and alsa-lib-devel. When I run a test on the drivers rpm, there are conflicts with the kernel headers already on the box. And I have no sound now, except in the audio configuration screen which is using the 14.rc3 packages. The packages are named with rh7 instead of fc7 as well. Do I have to run make install to get everything working, or is the header incompatability in the drivers a show stopper? Thanks.

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[alsa-devel] MOTU PCI-424
by Per Eriksson 14 Jul '07

14 Jul '07

Hello, After talking to Jonathan Woithe about the development done to support MOTUs product families, Jonathan and I have reached the conclusion that ALSA support for the PCI-424 series of cards would mean a great benefit for Linux users to get high quality audio at a fair price. The PCI-424 card is used to drive the following products: MOTU 2408mk3 MOTU HD192 MOTU 24I/O The work will start this autumn, as soon as I get everything set-up, and the progress will be published on https://erinet.se/alsa/motu_pci_424/pci-424 specification.pdf The actual driver will for me be a later issue to solve, I hope to gather some understanding by reading the list and look through the releases. However, I would like some more information on how the Matrix is supposed to be used in the wiki system. Are the vendor pages only for supported devices, or also for devices concidered for being supported? Thanks, Per

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[alsa-devel] [PATCH] Fix request_irq error handling
by Ash Willis 14 Jul '07

14 Jul '07

request_irq returns 0 on success and a negative code on failure. Many ALSA drivers incorrectly check for failure by testing for a positive code instead of a negative one. Signed-Off-By: Ash Willis <ashwillis(a)programmer.net> -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains

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[alsa-devel] [PATCH] Add quirks for HP dx2200 and dx2250
by Claudio Matsuoka 13 Jul '07

13 Jul '07

This patch adds quirks for HP dx2200 and dx2250. They use the standard configuration for 3-stack ALC862.

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[alsa-devel] Setting audio channels on ICH5 with SALSA
by Fausto Carvalho Marques SIlva 13 Jul '07

13 Jul '07

Setting audio channels on ICH5 with SALSA generates and: EINVAL Numbers of channels is 1. Does anyone have some ideias?? ( on ICH6 and ICH7 works well) thanks... Fausto Carvalho Marques Silva

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[alsa-devel] [PATCH] Add quirk for HP Samba (ALC888)
by Claudio Matsuoka 13 Jul '07

13 Jul '07

Hi, This patch fixes 5.1 surround output for the HP Samba using the same quirk as the HP Lucknow. If HP machines are uniform in their wiring, I think we should rename ALC888_HP_NETTLE to ALC888_6ST_HP and ALC888_HP_LUCKNOW to ALC888_3ST_HP for generic HP 6stack and 3stack configurations. I'll probably receive more HP machines soon, so I'll see if this is the case. Best regards, Claudio

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[alsa-devel] [PATCH] LG LW line-in capture
by Claudio Matsuoka 13 Jul '07

13 Jul '07

This patch adds line-in capture to LG LW (ALC880 codec). Tested with LW20.

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[alsa-devel] hda-intel on latext macbook pro
by Steffen Schulz 13 Jul '07

13 Jul '07

Hi all, I wrote about a week ago about the problems with alsa on the latest macbook pro (rev3, with santa-rosa). As I described, model=w2jc sort of works for the macbooks. Front Mic and LineIn work. The LineIn can be configured to LineOut by setting 2-Channe to 6-Channel. I've seen where this is done in the code and it is also possible to have a working Mic in this configuration. As I have no idea how the driver or auto-detection or driver development/reverse engineering usually works, I tried to brute force the mixer controls. I found two more mute switches, one for the front speakers and one for the LineOut. But I was unable to activate the normal headphone output. Is there anyone here who can give me a hint what to search for? I noticed that when the Mic is disabled with the code that switches between 2ch and 6ch, the Mic control do not work anymore. Maybe it is the same for the headphone controls and some other settings? This is what I got so far: [ALC882_MBP] = { .mixers = { alc882_mbp_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, .need_dac_fix = 1, .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, }, static struct snd_kcontrol_new alc882_mbp_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE ("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE ("Front Speaker Switch", 0x14, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE ("Line Out Switch", 0x1a, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE ("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), . . .[many controls that seem to do nothing] . { } }; I have no audio CD and no SPDIF-hardware, but the other controls do as they advertised. I would appreciate hints on finding the headphone controls. Hardware information can be seen in my last posts to the list. /Steffen -- Bildet Olsenbanden!

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Re: [alsa-devel] RE : RE : M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 13 Jul '07

13 Jul '07

> With the help of Hakan I made a fix and updated the doc accordingly. > Hakan could you test this fix (to be applied after my last one) and > report if it is ok for you ? Yes, seem to work quite nicely now with AC3/DTS also in device_setup=0x00 mode. /Håkan

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[alsa-devel] snd-hda-intel auto detect incorrect
by Matt Hirsch 12 Jul '07

12 Jul '07

I have an hp pavilion tx 1000 laptop running fedora 7. The auto detect feature of the snd-hda-intel dirver doesn't work on my sound card. As requested in Documentation/sound/alsa/ALSA-Configuration.txt I am forwarding the following to the developer mailing list: #uname -a Linux laptop 2.6.21-1.3228.fc7 #1 SMP Tue Jun 12 14:56:37 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux # lspci -s 00:10.1 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) # lspci -s 00:10.1 -nv 00:10.1 0403: 10de:026c (rev a2) Subsystem: 103c:30bf Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at c0000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping # yum list alsa-\* Installed Packages alsa-lib.x86_64 1.0.14-0.4.rc3.fc7 installed alsa-lib.i386 1.0.14-0.4.rc3.fc7 installed alsa-utils.x86_64 1.0.14-0.7.rc2.fc7 installed With no options I get the following log message and the audio doesn't work: kernel: hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from BIOS... When loading the module as follows, the audio works: modprobe snd-hda-intel model=3stack I believe the card was detected as 3stack-dig, as the alsamixer options match when either no option or model=3stack-dig are given. Hope this helped. Thanks for all your hard work! Matt

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[alsa-devel] Period Size in hw_params
by Paul Kavan 12 Jul '07

12 Jul '07

I finally have my basic audio applications working. I divided it into capture and playback and began with playback as I had a nice sinewave I could use to playback. Anyway, I think my hangup with getting the thing working was with the buffer and period information. My apps now do this: int periods = 2; /* Number of periods */ snd_pcm_uframes_t periodsize = 1000; /* Periodsize (bytes) */ /* Set number of periods */ if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return(-1); } /* Set buffer size */ size = periodsize * periods; if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (size)>>2) < 0) { fprintf(stderr, "Error setting buffer size.\n"); return(-1); } I borrowed the period size from using aplay -v and seeing what it used. Setting number of periods to 2 was a little bit of guess work based on what others seem to use. I have looked at the alsa documentation and cannot see how I could have determined these numbers would have worked without trial and error. Can anyone either give me a explanation of how to determine these or a pointer to some decent documentation that talks about this? Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] [PATCH] hda: stac9202 mixer fix
by Matthew Ranostay 12 Jul '07

12 Jul '07

Added support for line out controls for STAC9202 cards, and fixed issue where master mixer control was being created twice for headphone and speaker outs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 273fac42ced9 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Jul 04 15:17:38 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Wed Jul 11 22:13:06 2007 -0400 @@ -334,8 +334,6 @@ static struct snd_kcontrol_new stac9200_ }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -1401,7 +1399,15 @@ static int stac92xx_auto_create_hp_ctls( continue; add_spec_dacs(spec, nid); } - + for (i = 0; i < cfg->line_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); + } for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2",

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[alsa-devel] Can I mmap buffer allocated using SNDRV_DMA_TYPE_CONTINUOUS?
by Pharaoh . 12 Jul '07

12 Jul '07

Hi, I have allocated the DMA buffer using snd_pcm_lib_malloc_pages() and it was preallocated as: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data (GFP_KERNEL), 128 * 1024, 128 * 1024); Can I mmap such a dma buffer? I think yes... -pharaoh.

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[alsa-devel] [RFC] SGI O2 MACE audio ALSA module
by TJ 12 Jul '07

12 Jul '07

Hi, It's been a while since I last posted something. Here is the latest version of my code. It is somewhat of a mess at the moment, but I plan to tidy it all up when it works better, any // comments are not permanent features. I have been using trial and error to find why it play too fast. I am glad to say that it does play sound correctly, bar one little issue that has me stuck at the moment, it plays too fast (for any sample frequency). Also how fast appears to depend on the player (tried aplay and ogg123). I have cc the ALSA dev too in the hope that between the two lists someone may spot something. Please can anyone who replies cc me. The patch was built against (applies to) linux-2.6.19.7 from linux-mips.org. It'll probably work on other versions. Notes on the module it self... There are two code 'paths' depending on sma_indirect=1|0, this var will be made a module load argument. When 0 ALSA will be writing directly to the MACE ring buffer, currently the state of this code is believed very broken I have not been working on it for a while. When 1 ALSA will be writing to a bounce buffer and the module will copy data to the ring buffer as per the original driver. Currently sma_indirect is hard coded 1 and it is this 'path' i'm currently interested in fixing. The module has some proc files for debugging, the contents of the ad1843 regs, and for values in in mace. There is a _spy version of the module (which wont exist in my final patch) that provides just these proc files without making any (init) changes to any of the hardware (ie read only). This allowed me to see what state the O2 jingle left the various bit in. The module has a limited mixer, which I want to do a lot more with in future, and the front volume buttons are supported. DAC1/ADC map to dev0 on the card, while DAC2 maps to dev1. I don't know what else to say, please have a listen and a look. It will play any samples 8k to 48k, and any bit format that ALSA knows about. Regards, Thorben

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[alsa-devel] [PATCH] Make s3c24xx_i2s_set_clkdiv() change the correct bits
by Matt Reimer 12 Jul '07

12 Jul '07

Make s3c24xx_i2s_set_clkdiv() change the correct bits. Signed-off-by: Matt Reimer <mreimer(a)vpop.net> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 8ca314d..39f0246 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -344,11 +344,11 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, DBG("Entered %s\n", __FUNCTION__); switch (div_id) { - case S3C24XX_DIV_MCLK: + case S3C24XX_DIV_BCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; - case S3C24XX_DIV_BCLK: + case S3C24XX_DIV_MCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; -- 1.5.1.4

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[alsa-devel] [PATCH] Add LG LW20 si3054 modem id
by Claudio Matsuoka 12 Jul '07

12 Jul '07

This patch adds the id of the LG LW20 laptop modem.

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[alsa-devel] Using speaker-test for sound cards which dont support mmap
by Pharaoh . 12 Jul '07

12 Jul '07

Is it possible to use speaker-test for sound cards which do not support mmap? I straced speaker-test and it uses mmap call, can we bypass this?

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[alsa-devel] Help with ALSA application development
by Paul Kavan 12 Jul '07

12 Jul '07

Hello: I am still trying to write my own basic application that captures and plays audio. I am working in an embedded linux environment and have the drivers working well. I can use arecord/aplay without any problem. However, I have been trying to follow some web examples for simple capture and playback without any success. Does anyone know of some sample code for S8 formatted, 8k rate capture and playback. I think my problem is two-fold. First is getting the hw-params set correctly. I have a function call that sets the interface up. I am trying to use the same params that aplay -v returns to me, but perhaps I have them wrong somehow. Secondly is buffering the samples. I am not sure I have that correct. Does anyone have some pointers on that? I have attached my code. If anyone has some time and insight into this, I would greatly appreciate it. Thanks. Paul

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