November 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Invitation to connect on LinkedIn
by Arun KS 19 Nov '08

19 Nov '08

LinkedIn ------------ ALSA, I'd like to add you to my professional network on LinkedIn. - Arun Learn more: https://www.linkedin.com/e/isd/409983693/itycjb3I/ ------------------------------------------ What is LinkedIn and why should you join? http://learn.linkedin.com/what-is-linkedin/ ------ (c) 2008, LinkedIn Corporation

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Re: [alsa-devel] [RFC][PATCH] ELD routines and proc interface
by Wu Fengguang 19 Nov '08

19 Nov '08

On Tue, Nov 18, 2008 at 11:11:35PM -0800, Shane W wrote: > On Thu, Nov 13, 2008 at 10:21:53AM +0800, Wu Fengguang wrote: > > Create hda_eld.c for ELD routines and proc interface. > > Hi, Hi Shane, > Just testing this and your multichannel HDMI code over here > but it doesn't seem to get ELD data for my setup. Intel > g45-id board with an Yamaha RX-v1800 receiver which is HDMI > 1.3 capable. Do you run the latest xorg intel driver? Also dmesg should tell us something if anything goes wrong with ELD retrieving/parsing. > shane@chintoka:~$ cat /proc/asound/Intel/eld#3 > monitor name > connection_type HDMI > eld_version [0x0] reserved > edid_version [0x0] no CEA EDID Timing Extension > block present > manufacture_id 0x0 > product_id 0x0 > port_id 0x0 > support_hdcp 0 > support_ai 0 > audio_sync_delay 0 > speakers [0x0] > sad_count 0 > > 2ch audio is coming through just fine over HDMI though so > the connection itself is there. Yes. Obviously the ELD stuct has not been populated yet. Thanks, Fengguang

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[alsa-devel] [PATCH 0/4] multi-channel HDMI audio support
by Wu Fengguang 19 Nov '08

19 Nov '08

Hi Takashi, This patchset attempts to support multi-channel HDMI audio. It contains two basic operations: - setup the CA(channel-allocation) byte in the audio infoframe - setup the converter channel to HDMI slot mapping Please read patch 3 for more detailed descriptions: [PATCH 1/4] hda: make standalone hdmi_fill_audio_infoframe() [PATCH 2/4] hda: make global snd_print_channel_allocation() [PATCH 3/4] hda: HDMI channel allocations for audio infoframe [PATCH 4/4] hda: HDMI channel mapping cleanups The current implementation only does minimal works that ensure multi-channel audio playing out. However the wrong speakers may be selected. We could elaborate the framework/policy if it proves to be a problem. Thank you, Fengguang --

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[alsa-devel] [PATCH] hda: EAPD mute on suspend
by Matthew Ranostay 19 Nov '08

19 Nov '08

Moved support for EAPD mute on suspend from stac92hd71xx_suspend to the generic stac92xx_suspend function. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index e0298df..900d99f 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -4077,7 +4077,17 @@ static int stac92xx_resume(struct hda_codec *codec) codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); return 0; } -#endif + +static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct sigmatel_spec *spec = codec->spec; + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); + return 0; +} +#endif static struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, @@ -4087,6 +4097,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { .unsol_event = stac92xx_unsol_event, #ifdef SND_HDA_NEEDS_RESUME .resume = stac92xx_resume, + .suspend = stac92xx_suspend, #endif }; @@ -4498,14 +4509,8 @@ static int stac92hd71xx_resume(struct hda_codec *codec) static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) { - struct sigmatel_spec *spec = codec->spec; - - stac92hd71xx_set_power_state(codec, AC_PWRST_D3); - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); - return 0; + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + return stac92xx_suspend(codec, state); }; #endif

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[alsa-devel] HDA-emu v0.0.3
by Takashi Iwai 18 Nov '08

18 Nov '08

Hi, right now I updated hda-emu to version 0.0.3. hda-emu is a HD-audio emulation program that is hopefully useful for testing and debugging HD-audio drivers. This version contains a simple PCM test in addition to handling of control elements. The latest package is found at ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/misc/hda-emu-0.0.3.tar.gz The git tree is git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git Any fixes, improvements and feedbacks are appreciated thanks, Takashi

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[alsa-devel] [PATCH 0/17] ASoC updates
by Mark Brown 18 Nov '08

18 Nov '08

The following changes since commit 71cfc9028d762419ce4dea62b4afc9c32c4b4820: Mark Brown (1): ASoC: Add WM8728 codec driver are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Bryan Wu (1): ASoC: Blackfin: Simplify the MMAP_SUPPORT macros protected code Christian Pellegrin (2): ASoC: UDA134x codec driver ASoC: Machine driver for for s3c24xx with uda134x Cliff Cai (3): ASoC: AD1980 codec: add multi-channel function support ASoC: Blackfin: add multi-channel function support ASoC: Blackfin: Fix AD1980/1 build with MMAP support disabled Mark Brown (5): ASoC: Allow writes to uncached registers in WM8990 ASoC: Enable WM8990 ADC clocking workaround ASoC: Manage VMID mode for WM8990 ASoC: Build tlv320aic23 cleanly ASoC: Move uda134x_codec.h to uda134x.h Michael Hennerich (1): ASoC: Fix Blackfin AC97 DAI probe function return code Mike Frysinger (3): ASoC: Blackfin: updates Kconfig for SPORT ASoC: Blackfin: do not force TWI bus for ssm2602 codec ASoC: Blackfin: always set a default value for that GPIO range Naresh Medisetty (1): ASoC: DaVinci: Fix audio stall when doing full duplex Peter Ujfalusi (1): ASoC: Fix for master playback/capture volume range for TWL4030 codec include/sound/l3.h | 18 + include/sound/s3c24xx_uda134x.h | 14 + include/sound/uda134x.h | 26 ++ sound/soc/blackfin/Kconfig | 18 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 101 +++--- sound/soc/blackfin/bf5xx-ac97.c | 158 +++++---- sound/soc/blackfin/bf5xx-ac97.h | 35 ++- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ad1980.c | 19 +- sound/soc/codecs/l3.c | 91 +++++ sound/soc/codecs/tlv320aic23.c | 13 +- sound/soc/codecs/twl4030.c | 4 +- sound/soc/codecs/uda134x.c | 656 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda134x.h | 36 ++ sound/soc/codecs/wm8990.c | 23 +- sound/soc/codecs/wm8990.h | 4 +- sound/soc/davinci/davinci-i2s.c | 31 +- sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c24xx_uda134x.c | 374 ++++++++++++++++++++ 22 files changed, 1488 insertions(+), 154 deletions(-) create mode 100644 include/sound/l3.h create mode 100644 include/sound/s3c24xx_uda134x.h create mode 100644 include/sound/uda134x.h create mode 100644 sound/soc/codecs/l3.c create mode 100644 sound/soc/codecs/uda134x.c create mode 100644 sound/soc/codecs/uda134x.h create mode 100644 sound/soc/s3c24xx/s3c24xx_uda134x.c

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[alsa-devel] [PATCH 01/17] ASoC: Fix for master playback/capture volume range for TWL4030 codec
by Mark Brown 18 Nov '08

18 Nov '08

From: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> FGAIN for playback is in range of 0-0x3f, while for capture GAIN it is in the range of 0-0x1f. The original value of 128 (0x7f) would modify the CGAIN also for playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Steve Sakoman <steve(a)sakoman.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c1893d2..c778eb4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,10 +192,10 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), + 0, 0x3f, 0), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + 0, 0x1f, 0), }; /* add non dapm controls */ -- 1.5.6.5

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[alsa-devel] (no subject)
by Bryan Wu 18 Nov '08

18 Nov '08

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[alsa-devel] [PATCH 0/4] [RFC] multi-channel HDMI audio support
by Wu Fengguang 18 Nov '08

18 Nov '08

Hi all, This patchset attempts to support multi-channel HDMI audio. It contains two basic operations: - setup the CA(channel-allcation) byte in the audio infoframe - setup the converter channel to HDMI slot mapping Please check the last two patches for the detailed descriptions: [PATCH 1/4] hda: make standalone hdmi_fill_audio_infoframe() [PATCH 2/4] hda: make global snd_print_channel_allocation() [PATCH 3/4] hda: HDMI channel allocations for audio infoframe [PATCH 4/4] hda: HDMI channel mapping cleanups The current implementation only does minimal works that can ensure the multi-channel audio playing out. However the wrong speakers may be choosed, and I really need more information/knowledge/instructions from you to get it right. Thank you, Fengguang --

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[alsa-devel] 5d5d5f43f (Samsung hda-intel auto-sensing) causes input regression
by Daniel Drake 18 Nov '08

18 Nov '08

Hi, Just wanted to bring to your attention: Commit 5d5d5f43f1b835c375de9bd270cce030d16e2871 causes a regression on some laptops: input no longer works for some users. > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Feb 12 12:11:36 2008 +0100 > > [ALSA] hda-codec - Implement auto-mic jack sensing on Samsung laptops > > Implemented the auto-mic jack sensing for Samsung laptops with AD1986A > codec chip (model=laptop-eapd). > > The hardware uses pin 0x1d and 0x1f for the internal and external > mics, respectively. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Reported at: https://bugs.gentoo.org/show_bug.cgi?id=235732 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4228 The discussion around this patch appears to be here: http://thread.gmane.org/gmane.linux.alsa.devel/50989/focus=51457 Anything we can do to help solve this problem? Thanks, Daniel

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