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

[alsa-devel] [PATCH 1/2] ASoC Samsung S3c24xx updates - audio DMA cleanup.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch cleans up the audio DMA for the Samsung S3C24xx platform. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 3

[alsa-devel] [PATCH] [ALSA] Add first generation macbook subsystem ID
by bainonline＠gmail.com 17 Apr '07

17 Apr '07

From: Abhijit Bhopatkar <bainonline(a)gmail.com> First generation MacBooks were getting ignored by sigmatel drivers and wrongly being identified as MACMINI. This patch makes them identify as MACBOOK. Signed-off-by: Abhijit Bhopatkar <bainonline(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c94291b..cb99df5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1905,6 +1905,9 @@ static int patch_stac922x(struct hda_codec *codec) */ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id); switch (codec->subsystem_id) { + case 0x106b0a00: /* MacBook First generatoin */ + spec->board_config = STAC_MACBOOK; + break; case 0x106b0200: /* MacBook Pro first generation */ spec->board_config = STAC_MACBOOK_PRO_V1; break; -- 1.4.4.2

2 1

[alsa-devel] [1/2] 2.6.21-rc7: known regressions
by Adrian Bunk 17 Apr '07

17 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : ali_pata: boot from CD fails References : http://lkml.org/lkml/2007/3/31/160 Submitter : Stephen Clark <Stephen.Clark(a)seclark.us> Status : unknown Subject : kernels fail to boot with drives on ATIIXP controller (ACPI/IRQ related) References : https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=229621 http://lkml.org/lkml/2007/3/4/257 Submitter : Michal Jaegermann <michal(a)ellpspace.math.ualberta.ca> Status : unknown Subject : boot failure: rtl8139: exception in interrupt routine References : http://lkml.org/lkml/2007/3/31/160 Submitter : Stephen Clark <Stephen.Clark(a)seclark.us> Status : unknown Subject : laptops with e1000: lockups References : https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=229603 Submitter : Dave Jones <davej(a)redhat.com> Handled-By : Jesse Brandeburg <jesse.brandeburg(a)intel.com> Status : problem is being debugged Subject : forcedeth: interface hangs under load References : http://lkml.org/lkml/2007/4/3/39 http://www.spinics.net/lists/netdev/msg28981.html Submitter : Ingo Molnar <mingo(a)elte.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Ayaz Abdulla <aabdulla(a)nvidia.com> David Miller <davem(a)davemloft.net> Status : problem is being debugged Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available

5 4

[alsa-devel] [PATCH] ASoC export AC97 DAI
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch exports the ASoC AC97 Digital Audio Interface as a GPL'ed symbol. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 1

[alsa-devel] [PATCH 1/2] ASoC WM8753 codec support.
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch series adds support for the WM8753 codec as found on the OpenMoko Neo 1973 (other Neo 1973 and Samsung S3C24xx patches to follow today) as well other new devices. Features:- o HiFi and Voice DAI supported (inc runtime switching of DAI mode) o DAPM o All mixers o PLL calculator o 16,20 and 24bit samples. o WM8753 I2C ID added to include/linux/i2c-id.h Signed-off-by: Harald Welte <laforge(a)openmoko.org> Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Seth Forshee <seth.forshee(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

2 4

[alsa-devel] [PATCH 2/2] ASoC Samsung S3C24xx updates - i2s
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch adds DMA size fields to the S3C24xx audio DMA params and exports the S3C24xx I2S digital audio interface. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

1 0

Re: [alsa-devel] [PATCH] Re: iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 16 Apr '07

16 Apr '07

At Thu, 05 Apr 2007 10:47:52 +0200, Dominique Dumont wrote: > > Takashi Iwai <tiwai(a)suse.de> writes: > > > The patch is a bit dangerous because it unconditionally resets the > > DIGI_CONVERT verb for each widget. > > Agreed. We could also test the NID before switching on and off DigEn > in the DIGI_CONVERT. > > OTOH the switch would occur for *all* HDA chips which may not be what > you want. I'm not 100% sure that this particular procedure is necessary for all codecs. So, I prefer having it in the hda-codec part rather in hda-intel part. > > The below is my modified version. > > It doesn't turn off before the controller side reset as you did. So, > > if my version doesn't work, it implies that the SPDIF reset has to be > > done before the controller reset. Please give it a try. > > It works also ! I've tested AC3 and 44KHz. I cannot test 96KHz or > 196Khz (my amp is too old). Good to hear. > > + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ > > I had more the impression that the SPDIF stream is messed up if you > change the stream format while the digital converter is working... > > > > + if (codec->spdif_ctls & 1) > > Shouldn't you use AC_DIG1_ENABLE for a proper patch ? > > if (codec->spdif_ctls & AC_DIG1_ENABLE) > > > + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, > > + codec->spdif_ctls & 0xfe); > > likewise: codec->spdif_ctls & 0xff & ~AC_DIG1_ENABLE ? > > > + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); > > + /* turn on again (if needed) */ > > + if (codec->spdif_ctls & 1) > > if (codec->spdif_ctls & AC_DIG1_ENABLE) ? Yeah, that'd be better. I'll fix and commit to the HG tree later. thanks, Takashi

4 20

[alsa-devel] [PATCH 1/3] ASoC AT91xxxx - SSC port DSP support
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch series by Frank Madarino updates the AT91xxxx core to add DSP/PCM audio hardware formats. Changes:- o Rename at19-i2s.c -> at91-ssc.c o Rename at91-i2s.h -> at91-ssc.h o Add DSP hardware formats. o Rename various I2S labels to SSC Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 1

[alsa-devel] [PATCH 2/2] ASoC WM8753 codec - build changes
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch adds the WM8753 codec driver to the kernel build system. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH 3/3] ASoC AT91xxxx build
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch by Frank Madarino updates the AT91xxxx Makefile and Kconfig to build the renamed SSC files. Changes:- o Rename various i2s labels to ssc Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH 2/3] ASoC AT91xxxx eti B1 machine SSC changes
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch by Frank Madarino updates the eti B1 machine to use the newer AT91xxxx SSC core with the DSP/PCM audio hardware changes. Changes:- o #include "at91-ssc.h" instead of "at91-i2s.h" o Rename various I2S labels to SSC Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH] ASoC DAPM switching for reentrant codec paths.
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch fixes an issue whereby power was applied to any inactive analog path that would leave and reenter a codec (e.g. ACOP -> ACIN on WM8753). This change now checks for such paths and DAPM will power them down when not in use. Signed-off-by: Seth Forshee <seth.forshee(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 4

[alsa-devel] [PATCH] ASoC WM9712 kmemdup
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch creates the WM9712 codec register cache using kmemdup instead of doing a kzalloc followed by a memcpy. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

1 1

Re: [alsa-devel] 2.6.21-rc6, ASUS M2V 1.01G, ALSA detection/non-support
by Richard A Nelson 16 Apr '07

16 Apr '07

On Sun, 15 Apr 2007, Jiri Slaby wrote: > Date: Sun, 15 Apr 2007 23:45:29 +0200 > From: Jiri Slaby <jirislaby(a)gmail.com> > To: Richard A Nelson <cowboy(a)cavein.org> > Cc: linux-kernel <linux-kernel(a)vger.kernel.org> > Subject: Re: 2.6.21-rc6, ASUS M2V 1.01G, ALSA detection/non-support > > Cc: alsa-devel(a)alsa-project.org > > Richard A Nelson napsal(a): >> My son has an earlier version of the M2V & BIOS (0901) and his gets full >> 6ch >> sound. >> >> The 1.01G, and related BIOS (1304 or 1701) do not properly handle the >> onboard sound: >> hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from >> BIOS... > > What does your /proc/asound/version say? Both machines say: Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). The machines are running Debian Sid -- Rick Nelson Life'll kill ya -- Warren Zevon Then you'll be dead -- Life'll kill ya

1 0

Re: [alsa-devel] 2.6.21-rc6, ASUS M2V 1.01G, ALSA detection/non-support
by Jiri Slaby 16 Apr '07

16 Apr '07

Really Cc: alsa-devel(a)alsa-project.org Richard A Nelson napsal(a): > My son has an earlier version of the M2V & BIOS (0901) and his gets full > 6ch > sound. > > The 1.01G, and related BIOS (1304 or 1701) do not properly handle the > onboard sound: > hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from > BIOS... What does your /proc/asound/version say? > it seems to truely be a BIOS issue, as I don't see a difference in lspci > -vvxxx between our systems: > > 80:01.0 Audio device: VIA Technologies, Inc. VIA High Definition Audio > Controller (rev 10) > Subsystem: ASUSTeK Computer Inc. Unknown device 81e7 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Step ping- SERR- FastB2B- > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- > <TAbort- <MAbort- >SERR- <PERR- > Latency: 0, Cache Line Size: 64 bytes > Interrupt: pin A routed to IRQ 17 > Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA > PME(D0+,D1-,D2-,D3ho t+,D3cold+) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ > Queue=0/0 Enable- > Address: 0000000000000000 Data: 0000 > Capabilities: [70] Express Unknown type IRQ 0 > Device: Supported: MaxPayload 128 bytes, PhantFunc 0, > ExtTag- > Device: Latency L0s <64ns, L1 <1us > Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- > Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- > Device: MaxPayload 128 bytes, MaxReadReq 128 bytes > Link: Supported Speed unknown, Width x0, ASPM unknown, > Port 0 > Link: Latency L0s <64ns, L1 <1us > Link: ASPM Disabled CommClk- ExtSynch- > Link: Speed unknown, Width x0 > 00: 06 11 88 32 06 00 10 00 10 00 03 04 10 00 00 00 > 10: 04 c0 ff fb 00 00 00 00 00 00 00 00 00 00 00 00 > 20: 00 00 00 00 00 00 00 00 00 00 00 00 43 10 e7 81 > 30: 00 00 00 00 50 00 00 00 00 00 00 00 0a 01 00 00 > 40: 00 30 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > 50: 01 60 42 c8 00 00 00 00 00 00 00 00 00 00 00 00 > 60: 05 70 80 00 00 00 00 00 00 00 00 00 00 00 00 00 > 70: 10 00 91 00 00 00 00 00 00 00 30 00 00 00 00 00 > 80: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > 90: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > a0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > b0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > c0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > d0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > e0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > > ALSA, instead of seeing ALC660, on my box sees ALC660-VD, with very > little in the way of devices (only front output) - which does not work > either :( > > Stumped on where to go from here... regards, -- http://www.fi.muni.cz/~xslaby/ Jiri Slaby faculty of informatics, masaryk university, brno, cz e-mail: jirislaby gmail com, gpg pubkey fingerprint: B674 9967 0407 CE62 ACC8 22A0 32CC 55C3 39D4 7A7E

1 0

[alsa-devel] HDA Intel - Conexant CX20549 loaded but not working
by S Boucher 15 Apr '07

15 Apr '07

Using alsa-driver-1.0.14rc3 I have the following problems: - cannot record from internal mic (of an hp dv6245) - spdif not working - using jack I get lots of xruns (I don't get any xrun on the exact same system if I use an M-Audio Transit, so I'm assuming something ain't kosher in the conexant driver) - I was able to record cleanly from ext mic (although I had to select int mic) I'm with Ubuntu Feisty. N E Waze, if I can be of assistance let me know soon. The nice HP DV6245 laptop is heading back to the store in 1-2 days (nice system, except I don't have time to learn sign language :-)). __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[alsa-devel] Kernel crash with 2.6.20.x : snd-usb-audio implicated...
by Chris Rankin 15 Apr '07

15 Apr '07

Hi, The 2.6.20.x kernels all crash on my SMP-HT machine, and the oops report always shows that udev was trying to access an endpoint belonging to the snd-usb-audio driver. The frustrating part is that this particular endpoint shouldn't even exist! (It doesn't exist whenever I check the /sys/class/usb_endpoint directory.) Is there any way that userspace ("winecfg" in my case) can cause the snd-usb-audio driver to switch from (If# = 2, Alt = 0) to (If# = 2, Alt > 0) please? Because that is currently the only way that I can see this endpoint ever being created. Thanks, Chris --- bugme-daemon(a)bugzilla.kernel.org wrote: > Date: Sat, 14 Apr 2007 05:22:23 -0700 > From: bugme-daemon(a)bugzilla.kernel.org > To: rankincj(a)yahoo.com > Subject: [Bug 8198] udevd triggers oops when walking through /sys > > http://bugzilla.kernel.org/show_bug.cgi?id=8198 > > ------- Additional Comments From rankincj(a)yahoo.com 2007-04-14 05:22 ------- > Created an attachment (id=11144) > --> (http://bugzilla.kernel.org/attachment.cgi?id=11144&action=view) > dmesg output from 2.6.20.7 oops > > I don't know if this output is any more useful than any of the others. And I > can't see where the "single bit error" comes from when the output is full of > "use after free" poison. However, this much at least is different: > > EIP: [<6b6b6b6b>] 0x6b6b6b6b SS:ESP 0068:ec9eaf48 > <3>Slab corruption: start=eceea92c, len=512 > Redzone: 0x5a2cf071/0x5a2cf071. > Last user: [<c0226642>](kfree_skbmem+0xb/0x9c) > 080: 6b 6b 6b 6b 6a 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > Single bit error detected. Probably bad RAM. > Run memtest86+ or a similar memory test tool. > Prev obj: start=eceea720, len=512 > Redzone: 0x170fc2a5/0x170fc2a5. > Last user: [<f886eda5>](usb_create_ep_files+0xf1/0x2c2 [usbcore]) > 000: f8 fd 1b c2 e0 ee 4e f7 01 00 00 00 00 00 00 00 > 010: 30 a7 ee ec 30 a7 ee ec f4 91 1e c0 96 90 1e c0 > Next obj: start=eceeab38, len=512 > Redzone: 0x5a2cf071/0x5a2cf071. > Last user: [<c0226642>](kfree_skbmem+0xb/0x9c) > 000: 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > 010: 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > > Again, it's claiming that the last sysfs file accessed was the non-existent > /class/usb_endpoint/usbdev1.3_ep84/dev, which I simply cannot explain because > this endpoint belongs to If# = 2, Alt > 0: > > I: If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 90 Ivl=1ms > I: If#= 2 Alt= 2 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 46 Ivl=1ms > I: If#= 2 Alt= 3 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 25 Ivl=1ms > I: If#= 2 Alt= 4 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 16 Ivl=1ms > > Could the ALSA USB audio driver be changing the "Alt" on the fly somehow, > causing usbcore to need to create the extra endpoint in /sys? ___________________________________________________________ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.ht…

1 0

[alsa-devel] [2/3] 2.6.21-rc6: known regressions
by Adrian Bunk 14 Apr '07

14 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : USB: iPod doesn't work (CONFIG_USB_SUSPEND) References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Caused-By : Marcelo Tosatti <marcelo(a)kvack.org> commit 1d619f128ba911cd3e6d6ad3475f146eb92f5c27 Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged Subject : USB: Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Patch : http://lkml.org/lkml/2007/4/5/154 Status : patches available

2 1

[alsa-devel] function location?
by Ashlesha Shintre 13 Apr '07

13 Apr '07

Hi, I am trying to write an application to just play a wave file and record sound using ALSA -- I looked at the pcm_min.c file and tried to run it - I am using UBUNTU and 1.0.11-7ubuntu3 sound library called libasound2. I cant seem to find the pcm.c file which has all the snd_pcm_open and other functions used in the pcm_min.c file! I am using : /usr/include/alsa/asoundlib.h header -- but i m not sure where the functions defined in this header are -- and, find / -name pcm.c yields nothing --nor does find / -name snd_pcm_open Any pointers please! Thanks, Ashlesha.

6 7

[alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 13 Apr '07

13 Apr '07

Dear Alsa developers, I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. I want to use the /etc/asound.conf (.asoundrc) to split the card (that has 8 in and 8 out - but I use just 6 in/out), with minimal latency. I mean, I want to record and play (duplex) from/to each virtual card concurrently and from separate application instances. My app works with S16_LE, 8000, mono. Is the following asound.conf (.asoundrc) correct and efficient for this? I got spikes of cpu load (at 5-6%) on each virtual card any roughly 256 seconds when using a(play|record) to record or play from/to any channel, and the spikes cumulate with the number of virtual cards concurrently in use (eg. if I use 6 virtual cards concurrently I got spikes of 30% cpu load each 256 seconds). I need to create ctls? It would behave better if I use an asym to link toghether each couple of input and output? Any hints? Thanks a lot in advance, Giovanni that is the /etc/asound.conf I'm using: pcm.capt { type dsnoop ipc_key 223456 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.capt3 { type plug slave.pcm capt ttable.0.2 1 } pcm.capt4 { #used type plug slave.pcm capt ttable.0.3 1 } pcm.capt5 { type plug slave.pcm capt ttable.0.4 1 } pcm.capt6 { type plug slave.pcm capt ttable.0.5 1 } pcm.capt7 { type plug slave.pcm capt ttable.0.6 1 } pcm.capt8 { type plug slave.pcm capt ttable.0.7 1 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 }

2 1

[alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 13 Apr '07

13 Apr '07

Dear Alsa developers, I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. I want to use the /etc/asound.conf (.asoundrc) to split the card (that has 8 in and 8 out - but I use just 6 in/out), with minimal latency. I mean, I want to record and play (duplex) from/to each virtual card concurrently and from separate application instances. My app works with S16_LE, 8000, mono. Is the following asound.conf (.asoundrc) correct and efficient for this? I got spikes of cpu load (at 5-6%) on each virtual card any roughly 256 seconds when using a(play|record) to record or play from/to any channel, and the spikes cumulate with the number of virtual cards concurrently in use (eg. if I use 6 virtual cards concurrently I got spikes of 30% cpu load each 256 seconds). I need to create ctls? It would behave better if I use an asym to link toghether each couple of input and output? Any hints? Thanks a lot in advance, Giovanni that is the /etc/asound.conf I'm using: pcm.capt { type dsnoop ipc_key 223456 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.capt3 { type plug slave.pcm capt ttable.0.2 1 } pcm.capt4 { #used type plug slave.pcm capt ttable.0.3 1 } pcm.capt5 { type plug slave.pcm capt ttable.0.4 1 } pcm.capt6 { type plug slave.pcm capt ttable.0.5 1 } pcm.capt7 { type plug slave.pcm capt ttable.0.6 1 } pcm.capt8 { type plug slave.pcm capt ttable.0.7 1 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 }

1 0

[alsa-devel] Playback Devices in PREPARED State Throw Off POLLOUT Events?
by Jeff Rush 13 Apr '07

13 Apr '07

A question re normal internal operation of ALSA. I'm using the poll() interface to wait for a audio playback device to ask for more samples. The device is in the PREPARED state, not RUNNING state. At the beginning of my program, _before_ I write any samples to it, I'm getting repeated POLLOUT events, when I do not yet have data available to send. They seem to come frequent enough to load down my CPU, rather than every N ticks, so it seems. I thought a device was idle in the PREPARED state, and you had to either set the start conditions and write data, or explicitly invoke snd_pcm_start(). In my case, I wanted to use snd_pcm_start() once I had data to send, and also used the start conditions, set at 50%, to cause the audio output to them proceed to the speaker. The output snd_pcm_status_dump() right after exiting from the very first poll() ever in the program shows the following, and that I have a POLLOUT event pending. --- Status of Headphones Player --- state : PREPARED trigger_time: 0.000000 tstamp : 1176119509.658821000 delay : 0 avail : 2400 avail_max : 0 POLLOUT fired, calling ready_for_write() I suppose I could force it back to an earlier state like SND_PCM_STATE_OPEN or SND_PCM_STATE_SETUP, but that seems a bit harsh, since the snd_pcm_hw_params() takes it to the SND_PCM_STATE_PREPARED itself. Thanks for any tips, -Jeff

3 5

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 13 Apr '07

13 Apr '07

On Fri, 13 Apr 2007, Johannes Berg wrote: > On Thu, 2007-04-12 at 13:15 +0200, Rene Herman wrote: > > > > With Johannes don't know what kind of sound the chip produces using > > > OSS I'm stuck again. I have attached the samples to this e-mail. What > > > would be a reasonable debug step here? > > > > This message got stuck in the moderation queue due to being too big to > > pass through. The list limit is set at 60 kB and while that may be too > > low a limit, the 1.5 MB that was attached to this one was rather big. > > Too big for public lists. > > > > I approved it anyway, and then the attachment got scrubbed none the less > > by the list software it seems. > > I copied the files: > http://johannes.sipsolutions.net/files/stefan-noise.wav > http://johannes.sipsolutions.net/files/stefan-aplay.wav > > The interesting thing to note is that aoss and snd-pcm-oss have the > exact same effect, which is totally strange unless... hey! does aoss > maybe fall back to real oss when it is available? Point is the OSS modules aren't loaded :\ Stefan

1 0

[alsa-devel] rawmidi timestamps
by Florian 13 Apr '07

13 Apr '07

Hi, thanks for replying the "blocking read" question - it works fine with poll(). I'm using its timeout mechanism to bail out every 200msecs of inactivity. I wondered what snd_rawmidi_status_get_tstamp() is meant to return? Briefly checking out the driver midlevel, the timestamp is just ignored and setmem'ed to 0, or did I miss anything? Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

2 1

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 12 Apr '07

12 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hi, With Johannes don't know what kind of sound the chip produces using OSS I'm stuck again. I have attached the samples to this e-mail. What would be a reasonable debug step here? Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHgiuYH1+F2Rqwn0RCidgAJ44R70RyxNLoWOh681UGFbP+rrVFACfROLN bVFdMfHcIP4nLUDfaqrsbbQ= =MRRf -----END PGP SIGNATURE-----

2 1