April 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC: Correct DSP/TDM mode (for twl4030 4 channel mode)
by Peter Ujfalusi 06 Apr '09

06 Apr '09

Hello, I have now the implementation for the 4 channel support on the twl4030 codec driver (and also for the omap-mcbsp). It does work nicely, but I'm not sure what Interface format (DSP_A or DSP_B or TDM?) should I use for it. Can someone help me out on this one? The TPS65950 TRM is publicly available, which has exactly the same audio block: http://focus.tij.co.jp/jp/lit/ug/swcu050b/swcu050b.pdf Page 712 (TDM mode, Sample length 16 bits; Word length 16 bits) My bet goes to SND_SOC_DAIFMT_DSP_A, but I'm not 100% sure about it. Thank you, Péter

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[alsa-devel] [PATCH 0/5] ARM: DaVinci: ASoC: Add audio support for DM646x
by Naresh Medisetty 06 Apr '09

06 Apr '09

Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch set adds audio support for DM646x This series is dependant on the following 3 patches submitted to the DaVinci git: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x [PATCH] ARM: DaVinci: Fix the machine_is_davinci_dm6467_evm() error Patch 1: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 2: Adds McASP clock support for the two instances of mcasp casp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> on Monstavista 2.6.18 kernel.

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[alsa-devel] [PATCH] Add 92HD81B1C device id
by Matthew Ranostay 06 Apr '09

06 Apr '09

Added device id in struct for codec 92HD81B1C (0x111d76d5). Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b5e108a..b34d78b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4895,6 +4895,7 @@ again: switch (codec->vendor_id) { case 0x111d7604: case 0x111d7605: + case 0x111d76d5: if (spec->board_config == STAC_92HD83XXX_PWR_REF) break; spec->num_pwrs = 0; @@ -5707,6 +5708,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, -- 1.5.6.3

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[alsa-devel] [PATCH] snd-atmel-abdac: increase periods_min to 6 instead of 4
by Hans-Christian Egtvedt 06 Apr '09

06 Apr '09

This patch increases periods_min to 6 from 4, this will remove any hickups where the buffer is not filled fast enough from user space. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com> --- sound/atmel/abdac.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index aeda7cf..232de1e 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -176,7 +176,7 @@ static struct snd_pcm_hardware atmel_abdac_hw = { .buffer_bytes_max = 64 * 4096, .period_bytes_min = 4096, .period_bytes_max = 4096, - .periods_min = 4, + .periods_min = 6, .periods_max = 64, }; -- 1.5.6.3

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[alsa-devel] [PATCH] hda - enable SPDIF output for Intel DX58SO board
by Wu Fengguang 06 Apr '09

06 Apr '09

ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both. DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add 0x10 as slave output to the existing 0x06. Reported-by: Jeroen Van Breedam <jeroen.vanbreedam(a)sgr5.be> Tested-by: Jeroen Van Breedam <jeroen.vanbreedam(a)sgr5.be> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -8764,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_t {} }; +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + static hda_nid_t alc1200_slave_dig_outs[] = { ALC883_DIGOUT_NID, 0, }; @@ -8809,6 +8813,7 @@ static struct alc_config_preset alc883_p .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), .channel_mode = alc883_3ST_6ch_intel_modes, .need_dac_fix = 1,

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[alsa-devel] [PATCH] opl3sa2: add ZV port control
by Krzysztof Helt 06 Apr '09

06 Apr '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add ZV port control switch. This patch is done after solution given in the ALSA bug #2872 report. The patch resolves the issue. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opl3sa2.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index f395b0f..d161e74 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -477,6 +477,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1, OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1), OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1, db_scale_5bit_12db_max), +OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0), }; static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = { -- 1.5.2.2 ---------------------------------------------------------------------- Szukasz pieniedzy? Wez podwojny limit zadluzenia w koncie direct. >> http://link.interia.pl/f20a3

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[alsa-devel] ASoC updates for 2.6.30
by Mark Brown 06 Apr '09

06 Apr '09

The following changes since commit 1f2186951e02f2a5bcda9459f63136918932385a: Takashi Iwai (1): ALSA: Fix wrong pointer to dev_err() in arm/pxa2xx-ac97-lib.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.30 Philipp Zabel (2): ASoC: pxa-ssp: Use 16-bit DMA for magician stereo ASoC: Add Magician machine support sound/soc/pxa/Kconfig | 10 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 560 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa-ssp.c | 12 +- 4 files changed, 581 insertions(+), 3 deletions(-) create mode 100644 sound/soc/pxa/magician.c

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[alsa-devel] X-FI Datasheets Question
by daedra 05 Apr '09

05 Apr '09

I remember about a year ago James Dutton received the datasheets from creative for the X-FI. I was just wondering if he ever attempted to write a driver? I am just curious because now we have the source driver from creative, but I am hoping for some more features like HWMIX and Wavetable support for the X-FI. Any chance of these features in the future?

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[alsa-devel] Device creation order
by Lennart Poettering 05 Apr '09

05 Apr '09

Heya! PulseAudio listens for hotplugged audio devices via hal/udev. For each card ALSA creates a bunch of device nodes in /dev. Before PA can open the card it needs to make sure that all devices nodes of the card got properly created. I.e. to make sure that surround sound and SPDIF works it is not sufficient to wait until one PCM device is available, but instead that *all* devices that belong to the card are available, i.e. device nodes created with permissions and ACLs set up correctly. Unfortunately there is no signal from udev/hal/kernel that would tell me explicitly that all subdevices of a specific devices are completely enumerated and device files created for. I have been discussing this a bit with Kay Sievers and we came to the conclusion that a simple fix would be if we could rely that the control (i.e. /dev/snd/controlCxx) device is always the last device to be created for a card. Looking at the drivers this seems to be generally the case, so I was wondering if I can rely on this? Do all drivers behave like this? Do all drivers expose a control device? Can we officially declare it part of the userspace API that the control device is the last one to be created? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Mixer problem
by James Gardiner 05 Apr '09

05 Apr '09

I'm currently in the process of adding support for my HP dv4-1222nr laptop to the drivers. It has an IDT 92HD71B7 codec chip and no matter how hard I try, the mixer wants to default to the ATI SB one. I'm checking my /proc and dmesg and it's telling me the same thing. The weird thing is that using "model=hp-dv5" in my modprobe is fixing the 0x0d node speaker problem and my mods to get the digital microphone operative are working somewhat. However, the mixer in the sigmatel source shows that it's capable of adding dmic boost to the mixer controls. A printk I added in the source at the point where it would start processing the dmic is not even reaching the function. I tried adding my subsystem id to the PCI quirks and that didn't even help. I'm lost... Any ideas?

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