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

Re: [alsa-devel] [alsa-cvslog] alsa-kmirror: ALSA kernel mirror repository branch, master now at 9d46f4a919532c3f29a4ca1df3a4ce4686b11f37
by Thierry Vignaud 27 Jun '08

27 Jun '08

noreply-git(a)alsa-project.org writes: > Hello, > > This is an automated email from the git hooks/update script, it was > generated because a ref change was pushed to the repository. > > Updating branch, master, > via 9d46f4a919532c3f29a4ca1df3a4ce4686b11f37 (commit) > from fc5102d6c6520f48d39faf572881bc520c41aeb7 (commit) > > - Log ----------------------------------------------------------------- > commit 9d46f4a919532c3f29a4ca1df3a4ce4686b11f37 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Tue May 20 13:25:35 2008 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Tue May 20 13:25:35 2008 +0200 > > removed .hgtags can we make the log appears in subject instead of useless sha1 ID?

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[alsa-devel] Building trouble - redefinition of struct timeval
by Caloro Maurizio 27 Jun '08

27 Jun '08

Please i need help for following error message "redefinition of struct timeval - /include/global.h" the configure script run without any Error, if i compile "alsa-lib-1.0.17rc2" on I386 Netbsd 4.0 i receive folowing error break. "./configure --prefix=/usr --sysconfdir=/etc --build=i386-unknown-netbsdelf4.0 --host=i386-unknown-netbsdelf4.0 --target=i386-unknown-netbsdelf4.0 --with-pythonlibs=lpython2.5 --with-pythonincludes=-I/usr/include/python2.5 --enable-shared=no --enable-static=yes" & gmake gmake ... > Making all in controll gmake[1]: Entering directory `/usr/source/alsa-lib-1.0.17rc2/src/control' if /bin/ksh ../../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I../../include -I../../include -g -O2 -MT cards.lo -MD -MP -MF ".deps/cards.Tpo" -c -o cards.lo cards.c; \ then mv -f ".deps/cards.Tpo" ".deps/cards.Plo"; else rm -f ".deps/cards.Tpo"; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I../../include -I../../include -g -O2 -MT cards.lo -MD -MP -MF .deps/cards.Tpo -c cards.c -o cards.o In file included from ../../include/local.h:109, from control_local.h:22, from cards.c:35: ../../include/global.h:138: error: redefinition of 'struct timeval' ../../include/global.h:143: error: redefinition of 'struct timespec' gmake[1]: *** [cards.lo] Error 1 gmake[1]: Leaving directory `/usr/source/alsa-lib-1.0.17rc2/src/control' gmake: *** [all-recursive] Error 1 i have see that this its a possible handle with Posix_C_Source but i dont found the correct answer for me trouble.... Please for any Help i'am happy. thanks MC -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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[alsa-devel] [alsa-kernel][PATCH] ASoC: TLV320AIC3X: Add mixer control for ADC highpass filter
by Jarkko Nikula 27 Jun '08

27 Jun '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ sound/soc/codecs/tlv320aic3x.h | 2 ++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4f0bf26..d138306 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -223,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] = { "differential of HPROUT", "constant VCM", "single-ended", "differential of HPLCOM", "external feedback" }; static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; +static const char *aic3x_adc_hpf[] = + { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -232,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; #define LINE1R_ENUM 5 #define LINE2L_ENUM 6 #define LINE2R_ENUM 7 +#define ADC_HPF_ENUM 8 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -242,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; static const struct snd_kcontrol_new aic3x_snd_controls[] = { @@ -292,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Input */ SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), + + SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index c1dd1ac..e600946 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -37,6 +37,8 @@ #define AIC3X_ASD_INTF_CTRLB 9 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 +/* Audio codec digital filter control register */ +#define AIC3X_CODEC_DFILT_CTRL 12 /* ADC PGA Gain control registers */ #define LADC_VOL 15 -- 1.5.5.4

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[alsa-devel] [PATCH] Add another VIA audio chip to ac97 quirks list
by walt 27 Jun '08

27 Jun '08

Author: Walter Sheets <w41ter(a)gmail.com> Date: Thu Jun 26 10:39:19 PDT 2008 x86: Add VIA audio device #1841 to ac97_quirk list Signed-off-by: Walter Sheets <w41ter(a)gmail.com> --- Takashi, feel free to modify this in any way you need to. Thanks. diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b585cc3..aa5c2e0 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1746,6 +1746,12 @@ static struct ac97_quirk ac97_quirks[] = { }, { .subvendor = 0x1019, + .subdevice = 0x1841, + .name = "ECS K7VTA3", + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x1019, .subdevice = 0x0a81, .name = "ECS K7VTA3", .type = AC97_TUNE_HP_ONLY

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[alsa-devel] sw parameter boundary
by Norbert van Bolhuis 26 Jun '08

26 Jun '08

is the boundary sw parameter still used ? where is it for ? in the PCM kernel core (e.g. snd_pcm_playback_avail) it clearly serves as a pointers wrap point, but which one ? The one at the end of the DMA buffer or the one at the end of a process (virtual) address range (4 GB) ? -- This message has been scanned for viruses and is believed to be clean

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[alsa-devel] ECS K7VTA3: Mixer channels reversed
by walt 26 Jun '08

26 Jun '08

Hi list, I just replaced a mobo with an ECS K7VTA3 and I find that the Headphones control is now controlling the master channel and the Master control does nothing. I tried 'modprobe via82xx ac97_quirk=hp_only' but that makes the audio device unusable (although alsamixer still seems happy to set the individual channels). Any other quirks I should try? Here is some hardware info: ALSA sound/pci/via82xx.c:2409: via82xx: DXS white list for ESC K8 found 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) Subsystem: Elitegroup Computer Systems Unknown device 1841 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin C routed to IRQ 5 Region 0: I/O ports at dc00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: VIA 82xx Audio Kernel modules: snd-via82xx I'm running Linus' git kernel, current as of today, and alsa 1.0.16.

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[alsa-devel] seg fault in libasound
by Jerry Geis 26 Jun '08

26 Jun '08

I am running asterisk on debian Etch , i486 with math emulation in kernel. Verified by /boot/config-file. I am getting a seg fault in alsa. First I ran straight debian with alsa install. I got the same seg fault. Second I installed 1.0.16 and I still get the seg fault. Third I installed 1.0.17rc2 and I still get the seg fault. Posted to users first and stan suggested I post to devel. Below is the trace. I am only using the Console/Dsp from asterisk at this point. Which is the interface to the soundcard and alsa. What shall I do? Jerry -------------------------- Program received signal SIGABRT, Aborted. [Switching to Thread -1224033360 (LWP 2440)] 0xb7d77947 in raise () from /lib/tls/libc.so.6 (gdb) where #0 0xb7d77947 in raise () from /lib/tls/libc.so.6 #1 0xb7d790c9 in abort () from /lib/tls/libc.so.6 #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6 #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0, src_area=0x81cb7ec, src_offset=5223, samples=816, format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499 #4 0xb744b7b7 in snd_pcm_dsnoop_sync_ptr (pcm=0x81cb930) at pcm_dsnoop.c:77 #5 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cb930) at pcm.c:932 #6 0xb7422b55 in snd1_pcm_generic_hwsync (pcm=0x81cc1c8) at pcm_generic.c:143 #7 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc1c8) at pcm.c:932 #8 0xb742fb4d in snd_pcm_rate_hwsync (pcm=0x81cc3f8) at pcm_rate.c:624 #9 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc3f8) at pcm.c:932 #10 0xb7417817 in snd1_pcm_read_areas (pcm=0x81cc3f8, areas=0xb70a5b70, offset=0, size=160, func=0xb7421e10 <snd_pcm_mmap_read_areas>) at pcm.c:6376 #11 0xb742111f in snd_pcm_mmap_readi (pcm=0x81cc3f8, buffer=0xb74a5a40, size=160) at pcm_mmap.c:236 #12 0xb740fff4 in snd_pcm_readi (pcm=0x81cc038, buffer=0xb74a5a40, size=2440) at pcm_local.h:521 #13 0xb749579b in alsa_read (chan=0x8213f18) at chan_alsa.c:683 #14 0x080855b1 in __ast_read (chan=0x8213f18, dropaudio=0) at channel.c:2100 #15 0x08087725 in ast_channel_bridge (c0=0x8212858, c1=0x817cf68, config=0xb70a6eac, fo=0xb70a5fe8, rc=0xb70a5fe4) at channel.c:2380 #16 0xb7c75592 in ast_bridge_call (chan=0x8212858, peer=0x8213f18, config=0xb70a6eac) at res_features.c:1490 #17 0xb75c38fd in dial_exec_full (chan=0x8212858, data=0xb70a8fd8, peerflags=0xb70a6f84, continue_exec=0x0) at app_dial.c:1773 #18 0xb75c519d in dial_exec (chan=0x8212858, data=0xb70a8fd8) at app_dial.c:1827 #19 0x080c6e3f in pbx_extension_helper (c=0x8212858, con=0x0, context=0x82129d8 "smvoice", exten=0x8212a28 "s", priority=3, label=0x0, callerid=0x8215450 "404", action=E_SPAWN) at pbx.c:537 #20 0x080c9185 in __ast_pbx_run (c=0x8212858) at pbx.c:2317 #21 0x080c9ebe in pbx_thread (data=0x8212858) at pbx.c:2636 #22 0x080f51d2 in dummy_start (data=0x8213178) at utils.c:909 #23 0xb7eff0bd in start_thread () from /lib/tls/libpthread.so.0 #24 0xb7e1b01e in clone () from /lib/tls/libc.so.6

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[alsa-devel] [asoc-dev][RFC/PATCH 0/2] Generic DAPM register modifier widget
by Jarkko Nikula 26 Jun '08

26 Jun '08

Hi At least tlv320aic3x codec has several use cases where multiple register bits are needed to change when widget power state changes and where these are not worth/safe to expose into user space. Like mic bias voltage. Currently there is one use case for this multiple bit change in tlv320aic3x: SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), There are two problems with that: 1. Two widgets having the same name and thus snd_soc_dapm_add_route will add only first one them into audio path 2. Even then bits aren't modified at once and thus some fault bit configuration might be present while power state is changing. So I came with this idea of generic register modifiers widget and event handler for it. -- Jarkko

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[alsa-devel] No sound on MacBook 3.1
by Marco Vermeulen 26 Jun '08

26 Jun '08

Hi there, I hope that someone can help me, as I've tried everything conceivable to get sound working on my Macbook 3.1. I've visited many a forum, and eventually came across this list. My last hope :) I purchased a Macbook 3.1 earlier this year. On installing Linux, I haven't been able to get any sound going. I've tried several distributions: Ubuntu Hardy, Fedora 8/9, and now openSuse. I came across a script that uploads local relevant information to a paste bin, so here's a link: http://pastebin.ca/1055610 Please help, I'm desperate to get this going on my machine! Thanks & Regards, Marco.

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[alsa-devel] support for RME HDSPe RayDAT
by Matthijs ten Berge 26 Jun '08

26 Jun '08

Hello all, <intro:> As you might already know, RME has released a successor for the HDSP 9652, the HDSPe RayDAT (http://www.rme-audio.de/en_products_hdspe_raydat.php). It is based on the 9652's design, with some new/changed features (e.g. more channels and lower latencies), but the most important hardware change is that it is now a PCI-express card. I'm currently in the process of choosing which audio interface to buy, the 'old' HDSP 9652 or the new RayDAT (I haven't found any other card with 3 or more ADAT in/outs in the same price range or below, but suggestions are welcome). Of course I can stay on the safe side and opt for the HDSP 9652, which has stable and complete driver support (right, eh?). But on the other hand the RayDAT has better features and future, but without the guarantee that it is or will be supported it can only serve as a very costly paper weight (and not even a good one I suppose). I wrote an e-mail to RME, but a bit too early, the card was not released at that moment. See below for their reply. <question:> Are there any plans to add support for the RME RayDAT to the HDSP driver/tools? Or perhaps somebody is already working on it right now? It is not mentioned in the Wiki matrix atm. I wish I could just do it myself, but (concerning kernel programming) I never did more than a 'hello world' kernel module, so I don't think this would be a good idea... (I do have coding experience, but not in-kernel).

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