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Re: [alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 17 Apr '07

17 Apr '07

I applied the first hint from Jaroslav (at the end of this post the new asound.conf), but the problem seems not gone: one time each 256 seconds I get a very strong cpu load spike, that is maybe related (my guess) to the usage of plugin (maybe periodic cleaning each 256 secs?). When there is no spike, the cpu load is 0.0% with occasional 0.1% any 60-90 secs. I ran the latency test code with plughw:0 (I can't use latency.c with my virtual cards because they can't link the in and out togheter) using the blocking mode, real time, no poll. Any 256 secs, spike! (a cpu load for latency.c of 45.00%, for roughly a sec, with the cumulated cpu time of the latency.c app reported from top -d 1.0 passing from: 0:00.01 to: 0:00.64 to: 0:01.26 etc, any 256 seconds. latency.c output: ./latency -P plughw:0 -C plughw:0 -f S16_LE -c 1 -r 8000 -m 320 -M 960 -s 600 -b Playback device is plughw:0 Capture device is plughw:0 Parameters are 8000Hz, S16_LE, 1 channels, blocking mode Wanted tick time: 0us, poll mode: no Loop limit is 4800000 frames, minimum latency = 320, maximum latency = 960 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 2 <- 0 3 <- 0 4 <- 0 5 <- 0 6 <- 0 7 <- 0 8 <- 0 9 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 10 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 + 5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 + 10*0.0833333 + 11*0.0833333 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 12 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Trying latency 320 frames, 40000.000us, 40.000000ms (25.0000Hz) Success Playback: *** frames = 4800320 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482713139 delay : 317 avail : 3 avail_max : 192 Capture: *** frames = 4800000 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482757926 delay : 1 avail : 1 avail_max : 189 Maximum read: 160 frames Maximum read latency: 20000.000us, 20.000000ms (50.0000Hz) Hardware sync Playback time = 1176471431.639255, Record time = 1176471431.639255, diff = 0 new asound.conf: pcm.c3 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 2 } } pcm.capt3 { type plug slave.pcm c3 } pcm.c4 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 3 } } pcm.capt4 { type plug slave.pcm c4 } pcm.c5 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 4 } } pcm.capt5 { type plug slave.pcm c5 } pcm.c6 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 5 } } pcm.capt6 { type plug slave.pcm c6 } pcm.c7 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 6 } } pcm.capt7 { type plug slave.pcm c7 } pcm.c8 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 7 } } pcm.capt8 { type plug slave.pcm c8 } ############################# ############################# 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 } On 4/13/07, Giovanni Maruzzelli <gmaruzz.lists(a)gmail.com> wrote: > Thanks a lot Jaroslav, I'll apply your suggestions and I'll report here > > On 4/13/07, Jaroslav Kysela <perex(a)suse.cz> wrote: > > On Fri, 13 Apr 2007, Giovanni Maruzzelli wrote: > > > > > 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? > > > > Use dsnoop bindings rather than routing (ttable) for capture too, > > otherwise alsa-lib works with all channels internally. > > > > > 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 > > > > aplay|arecord is not very good test. Use latency test code - available in > > alsa-lib/test directory. > > > > Jaroslav > > > > ----- > > Jaroslav Kysela <perex(a)suse.cz> > > Linux Kernel Sound Maintainer > > ALSA Project, SUSE Labs > > >

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[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>

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[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

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[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

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[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>

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[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

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[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>

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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

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[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>

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[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>

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