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[alsa-devel] Snd problem ... ( ACCESS_TYPE )
by errordeveloper＠gmail.com 03 Oct '07

03 Oct '07

Hi list. I' m using alsa-lib v1.0.14rc2 on gentoo-linux (v2.6.20-gentoo-r8) my soundcard is RME Hammerfall HDSP9632 (snd-hdsp driver) i have been using it for a while - all fine i only run a few applications for my sound experiments. one of those is Snd (from CCRMA), it was okay until i have done a few updates lately - kernel from 2.6.17 to 2.6.20 and alsa-lib as well. i'm not sure about either of these but i check Snd src-tree and the mailing list - nothing was changed .. the error msg that i get is: :(play) ;status of hw:0 hw_params status of hw:0 ACCESS: MMAP_NONINTERLEAVED RW_NONINTERLEAVED FORMAT: S32_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: [128 384] CHANNELS: [4 12] RATE: [32000 192000] PERIOD_TIME: (333 256000] PERIOD_SIZE: [64 8192] PERIOD_BYTES: [2560 393216] PERIODS: 2 BUFFER_TIME: (666 512000] BUFFER_SIZE: [128 16384] BUFFER_BYTES: [2560 786432] TICK_TIME: 1000 sw_params status of hw:0 start_mode: DATA xrun_mode: STOP tstamp_mode: NONE period_step: 0 sleep_min: 0 avail_min: 0 xfer_align: 0 silence_threshold: 0 silence_size: 0 boundary: 0 ;Invalid argument: hw:0: access type RW_INTERLEAVED not available can't play : Invalid argument: hw:0: access type RW_INTERLEAVED not available #f : i had tried to modify audio.c in Snd's tree to use both ACCESS types given above , but it didn't help - showed device is busy error instead. i'm only studying C, so can't be sure .. anyway here is a problem which i could not solve yet and haven't seen anyone helpful around #alsa, neither i could google anything out. for a ref.: Cmdist mailing list Cmdist(a)ccrma.stanford.edu http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist and our discussion is here: http://ccrma-mail.stanford.edu/pipermail/cmdist/2007-July/004221.html could anyone have a look at this at somepoint, please . -- thank you, Ilya. D errordeveloper|*@*|gmail|*dot*|com

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[alsa-devel] ASOC: Best way to get private data from machine driver to the other drivers?
by Timur Tabi 02 Oct '07

02 Oct '07

I'm writing an ASOC driver. When my machine driver loads, it determines some hardware information for the DMA controllers, some other hardware information for the I2S controller, and even more information for the codec. I want to be able to pass all this information to each of the three other drivers. Unfortunately, it doesn't appear that I have enough "private_data" pointers available to me. The current ASOC drivers all use statically defined global structures and indexes into arrays of these structures to identify individual devices, such associating I2S controller #2 with DMA channels 3 and 4. On PowerPC, this doesn't work. The actual hardware configuration is determined at runtime (see arch/powerpc/boot/dts/), and so the machine driver needs to collect all the information and make all the device-to-device associations. For example, the PCM driver has an open function: static struct snd_pcm_ops fsl_dma_ops = { .open = fsl_dma_open, ... static int fsl_dma_open(struct snd_pcm_substream *substream) { ... What I would like is for fsl_dma_open() to be able to obtain pointers to the DMA controller's registers (there are two controllers with 4 channels each) by looking inside 'substream'. How can I do that? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] some problem about alsa lib usage on arm926
by Akio 02 Oct '07

02 Oct '07

Dear All: right now, I'm using alsa lib interface to work with voip application. Our hardware platform is an ARM926 266MHz box with Linux kernel 2.6.19.2. We got some problems. would you like to do us favors? 1. what is the different between "plughw:0,0", "defalut", "hw:0.0"? after reading the document in http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html about "PCM naming conventions", we still not very sure the relationship between those device conventions. is there any more detail documents talk about on this? right now, we are using alsa lib 1.0.13 and the device is "plughw:0,0". 2. Is there any documents detail discuss the setting about sample rate, buffer size, period size. because, I have two different voice codec to handle, one is narrow band(8k Hz), the other is wide band(16K Hz). right now, I using the following functions and setting to make alsa work. snd_pcm_hw_params_set_rate_near sets sample rate, 8k or 16k. snd_pcm_hw_params_set_period_size_near sets period size, current is 128. snd_pcm_hw_params_set_buffer_size_near sets buffer size, 8k is 2048, 16k is 4096. the unit seems be samples not byte. 3. the code was based on the following URL. http://www.suse.de/~mana/alsa090_howto.html http://www.equalarea.com/paul/alsa-audio.html open capture and playback at one function, and provide one read and one write interface to up layer. here comes three problems: a. the channel opened is just only one, but If I set the snd_pcm_hw_params_set_access as SND_PCM_ACCESS_RW_NONINTERLEAVED. using snd_pcm_readn and snd_pcm_writen the code will not work. what is the wrong? but if set snd_pcm_hw_params_set_access as SND_PCM_ACCESS_RW_INTERLEAVED and using snd_pcm_readi and snd_pcm_writei. the code work very well. one channel also have data interleave problem? I don't understand why? b. in snd_pcm_open the block flag is set as 0. what is the mode of the operation? blocking or non-blocking? c. call the xrun_recovery() timing. currently the xrun_recovery only do on EPIPE. the following is the handling core. err = snd_pcm_prepare(handle); if(err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) { nanosleep(&retryTime, &tmpTime); LOG("audio device resume fail, %s\n", snd_strerror(err)); } if(err < 0) { err = snd_pcm_prepare(handle); if(err < 0) LOG("audio data IO fail, %s\n", snd_strerror(err)); } } else { if(err < 0) LOG("audio device prepare fail, %s\n", snd_strerror(err)); } Does the code have any missing point to handle? If any missing, what should I to add? 4. BTW, I call xrun_recovery both in capture and playback direction. After long time test. the test procedure is as the following steps: 1. capture voice data as PCM data. 2. encoded as amr-wb bit stream. 3. decoded as PCM data from encoded bit stream. 4. playback PCM data. after 16 hours continue tests, there are some delay, about 800 mili seconds. is there any way to reduce the delay after long time test. we want make the delay about < 400 mili seconds after one week continue test. BTW, is there any way to prevent xrun_recovery calling? is there any index to monitor ALSA device buffer information? 5. If we want using salsa lib to replace the usage of alsa lib, what do we have to take care? how about the latency and performance between salsa and alsa? how about the stability of salsa? There are so many problems after many tests. If you found any documents we need to study more detail. please tell us. Thanks in advance. Thank you. Best Regards, Akio

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[alsa-devel] issue with ADI A1986A
by l1983＠bankerinter.net 01 Oct '07

01 Oct '07

Hello, I need some help from you guys. I,m a new linux user and totally unfamiliar with all this coding... I have an issue with my motherboard "ASSUS M2N-MX" and it's onboard sound chip :"ADI AD1986A" witch supports 6 channel CODEC , Jack-Sensing & Enumeration technology , and with S/PDIF OUT interface. Can you please somehow, (or somebody you know) , make for me some installable .deb archive that will help me to set up my sound, since i can't do it on my own. :( Or help me some how else, (either sending me some address's or by giving me some other advice. HELP ME PLEASE!!!!!! l1983(a)bankerinter.net

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[alsa-devel] short writes?
by william estrada 30 Sep '07

30 Sep '07

Hi group, I have written my own ALSA recorder and player. Along with that I have several bugs. The bug I am working on right now is 'short writes' when I use the pipe command into my player. My player runs and asks for the file to play, so that I may play incoming files as they are sent in. I have a program that uses 'inotify' to watch for incoming file and then pipe the file name into my player. Now with all of that, is there a API call that will put the writei into blocking mode? Or an API call that will flush the output from writei? Or any other suggestions? Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] nforce ac97 plays too fast
by Stas Sergeev 30 Sep '07

30 Sep '07

Hello. Trying to get a software amplification to work with NForce CK804 AC'97, I've found out that whatever the asound.conf I write, the sound starts playing much faster than it should. After some debugging, I've found out the following: - The card (or the driver) only supports 48000KHz. - The program (any, because libao and artsd suffer from that) is trying to set some rate with snd_pcm_hw_params_set_rate_near(), and that succeeds, but the returned rate is 48000, while the requested is, say, 22050. No wonder it then plays with the wrong speed. Of course the program can check the returned rate, but it sets the "direction" parameter to 0, which should mean "take exactly that value". I've looked into an alsa-tutorial, and the examples there do not check the returned rate too - dir==0 should be enough. Looking into the sources, it seems the culprit is the snd_interval_refine_min/max(), which do not care about the (i->min > min && !openmin) case at all. The question is: is setting dir=0 (or even NULL, since it is a pointer) enough to set the exact rate? And if so - how snd_interval_refine_min() should handle the (i->min > min && !openmin) case? Or if not - what does the dir==0 mean and why it is not adjusted to the proper value on return, when the rate is changed? (i.e. the rate parameter changes from 22050 to 48000, but the dir parameter remains 0, doesn't change) With the default configuration the rate is proper, but I find it annoying that altering asound.conf in pretty much any way gets the wrong results.

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[alsa-devel] Asus M2NPV-VM snd_hda_intel problem - also with Alsa 1.0.15rc3
by Ma Begaj 29 Sep '07

29 Sep '07

Hi, for the sweet love of ... does anyone have an idea how to make this sound chip working... I have Asus M2NPV-VM with AD1986A sound chip. I tried "model=3stack position_fix=1", but it does not help. surround does not work and sound mutes whenever I change volume. when I use model=ultra or eapd-laptop, stereo is at stable and I can change volume. but surround I tried it with vanilla kernel 2.6.21.5/26.22.6 and snd_hda_intel from the kernel without any success. I even compiled Alsa 1.0.15rc3, but situation did no change. anybody ... with any idea ... I would appreciate any help. any help at all. thanks. M.

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[alsa-devel] Alsa 1.0.15rc2 and oops with Dell M1210
by Herton Ronaldo Krzesinski 29 Sep '07

29 Sep '07

Hi, I have some reports of problems with Alsa 1.0.15rc2 and Dell M1210 (dell-m82 model). The following oops happens: Sep 27 19:28:34 dru kernel: Unable to handle kernel paging request at ffff81107b70f3a0 RIP: Sep 27 19:28:34 dru kernel: [<ffffffff88108751>] :snd_hda_intel:snd_hda_input_mux_info+0x31/0x50 Sep 27 19:28:34 dru kernel: PGD 8063 PUD 0 Sep 27 19:28:34 dru kernel: Oops: 0000 [1] SMP Sep 27 19:28:34 dru kernel: CPU 0 Sep 27 19:28:34 dru kernel: Modules linked in: ipv6 video thermal sbs fan container snd_seq_dummy button snd_seq_oss snd_seq_midi_event snd_seq do ck battery ac ide_cd binfmt_misc loop ntfs nls_utf8 nls_cp437 vfat fat dm_mirror dm_mod usb_storage ide_core hidp l2cap usbmouse usbhid ff_memless usbkbd hci_usb bluetooth cpufreq_ondemand cpufreq_conservative cpufreq_powersave acpi_cpufreq freq_table processor nvram pcspkr snd_usb_lib snd_r awmidi snd_seq_device b44 snd_pcm_oss snd_mixer_oss ssb pcmcia pcmcia_core uhci_hcd ehci_hcd usbcore mii ohci1394 ieee1394 i2c_i801 i2c_core evdev snd_hda_intel snd_pcm snd_timer snd_page_alloc snd_hwdep joydev tsdev mmc_block sdhci mmc_core snd soundcore iTCO_wdt iTCO_vendor_support sg sr_m od ext3 jbd scsi_wait_scan sd_mod ahci ata_piix libata scsi_mod Sep 27 19:28:34 dru kernel: Pid: 3466, comm: alsactl Not tainted 2.6.22.9-laptop-1mdv #1 Sep 27 19:28:34 dru kernel: RIP: 0010:[<ffffffff88108751>] [<ffffffff88108751>] :snd_hda_intel:snd_hda_input_mux_info+0x31/0x50 Sep 27 19:28:34 dru kernel: RSP: 0018:ffff81007aa65d48 EFLAGS: 00010206 Sep 27 19:28:34 dru kernel: RAX: 0000000ffffffff0 RBX: ffff81007b734d80 RCX: 00000000ffffffff Sep 27 19:28:34 dru kernel: RDX: 0000000000000000 RSI: ffff81007aa65d88 RDI: ffff81007aa65de0 Sep 27 19:28:34 dru kernel: RBP: ffff81007aa65d88 R08: ffff81007b70f3a8 R09: 000000000000000a Sep 27 19:28:34 dru kernel: R10: 0000000000000000 R11: 0000000000000246 R12: ffff81007c44a968 Sep 27 19:28:34 dru kernel: R13: ffff81007c44a800 R14: ffff81007adbaf40 R15: 00007fff95ccbdb0 Sep 27 19:28:34 dru kernel: FS: 00002ad615cc6e50(0000) GS:ffffffff8053e000(0000) knlGS:0000000000000000 Sep 27 19:28:34 dru kernel: CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b Sep 27 19:28:34 dru kernel: CR2: ffff81107b70f3a0 CR3: 000000007a926000 CR4: 00000000000006e0 Sep 27 19:28:34 dru kernel: Process alsactl (pid: 3466, threadinfo ffff81007aa64000, task ffff81007c659860) Sep 27 19:28:34 dru kernel: Stack: 00007fff95ccbdb0 ffffffff880b3835 0000000000000000 00007fff95ccbdb0 Sep 27 19:28:34 dru kernel: 00000000c1105511 ffff81007c44a800 ffff81007adbaf40 ffffffff880b4d88 Sep 27 19:28:34 dru kernel: 0000000000000001 0000000000000000 0000000000000000 0000000000000000 Sep 27 19:28:34 dru kernel: Call Trace: Sep 27 19:28:34 dru kernel: [<ffffffff880b3835>] :snd:snd_ctl_elem_info+0x55/0x150 Sep 27 19:28:34 dru kernel: [<ffffffff880b4d88>] :snd:snd_ctl_ioctl+0x648/0x8b0 Sep 27 19:28:34 dru kernel: [<ffffffff802aa045>] do_ioctl+0x35/0xe0 Sep 27 19:28:34 dru kernel: [<ffffffff802aa310>] vfs_ioctl+0x220/0x2c0 Sep 27 19:28:34 dru kernel: [<ffffffff802aa445>] sys_ioctl+0x95/0xb0 Sep 27 19:28:34 dru kernel: [<ffffffff8020a28e>] system_call+0x7e/0x83 Sep 27 19:28:34 dru kernel: Sep 27 19:28:34 dru kernel: Sep 27 19:28:34 dru kernel: Code: 49 8b 74 00 08 e8 95 bd 21 f8 31 c0 48 83 c4 08 c3 0f 1f 80 Sep 27 19:28:34 dru kernel: RIP [<ffffffff88108751>] :snd_hda_intel:snd_hda_input_mux_info+0x31/0x50 Sep 27 19:28:34 dru kernel: RSP <ffff81007aa65d48> Sep 27 19:28:34 dru kernel: CR2: ffff81107b70f3a0 I have a better report on a i386 kernel, unfortunately it's tainted: BUG: unable to handle kernel NULL pointer dereference at virtual address 00000000 printing eip: f8f89ed8 *pde = 00000000 Oops: 0000 [#1] SMP Modules linked in: snd_usb_audio snd_usb_lib snd_rawmidi snd_seq_device iwl3945 mac80211 cfg80211 b44 mmc_block nvidia(P) snd_hda_intel snd_pcm ohci1394 snd_timer ssb pcmcia sdhci pcmcia_core mii ieee1394 mmc_core joydev ipw3945 ieee80211 ehci_hcd snd_page_alloc snd_hwdep uhci_hcd ieee80211_crypt snd intel_agp i2c_i801 tsdev usbcore soundcore sg agpgart evdev i2c_core iTCO_wdt iTCO_vendor_support sr_mod reiserfs scsi_wait_scan sd_mod ahci ata_piix libata scsi_mod CPU: 0 EIP: 0060:[<f8f89ed8>] Tainted: P VLI EFLAGS: 00010246 (2.6.22.7-laptop-1mdv #1) EIP is at snd_hda_input_mux_info+0x38/0x50 [snd_hda_intel] eax: f6764a4c ebx: 00000000 ecx: ffffffff edx: f6269de4 esi: 00000000 edi: f6269e3c ebp: f6269db0 esp: f6269da4 ds: 007b es: 007b fs: 00d8 gs: 0033 ss: 0068 Process alsactl (pid: 2318, ti=f6268000 task=c237a550 task.ti=f6268000) Stack: f66da5c0 00000000 f6269de4 f6269db8 f8f995e4 f6269f48 f8edd1eb 00000000 bf979c70 f64fcac0 f66b6c00 f66b6d4c f8cf0000 f66b6d4c c0199456 00000000 00000001 00000002 00000000 00000000 75706e49 6f532074 65637275 00000000 Call Trace: [<c010529a>] show_trace_log_lvl+0x1a/0x30 [<c010535b>] show_stack_log_lvl+0xab/0xd0 [<c0105551>] show_registers+0x1d1/0x2d0 [<c0105768>] die+0x118/0x240 [<c0121399>] do_page_fault+0x199/0x630 [<c0313e6a>] error_code+0x72/0x78 [<f8f995e4>] stac92xx_mux_enum_info+0x14/0x20 [snd_hda_intel] [<f8edd1eb>] snd_ctl_ioctl+0x80b/0xc30 [snd] [<c018f308>] do_ioctl+0x28/0xb0 [<c018f5af>] vfs_ioctl+0x21f/0x2a0 [<c018f689>] sys_ioctl+0x59/0x70 [<c01041ba>] sysenter_past_esp+0x6b/0xa1 ======================= Code: 08 8d 7a 58 89 1c 24 c7 42 40 03 00 00 00 c7 42 48 01 00 00 00 8b 08 89 4a 50 8b 4a 54 8b 18 8d 73 ff 39 d9 0f 43 ce 8b 74 c8 04 <ac> aa 84 c0 75 fa 8b 1c 24 31 c0 8b 74 24 04 8b 7c 24 08 89 ec EIP: [<f8f89ed8>] snd_hda_input_mux_info+0x38/0x50 [snd_hda_intel] SS:ESP 0068:f6269da4 Accordingly to one of the reporters, when changing the model to 5stack the oops doesn't happens anymore and everything works flawlessly. At first I thought that the problem could be the different pin cfgs of dell-m82 compared to the old and replaced dell model. I was in contact with one of the reporters, but unfortunately reverting only the pin cfg didn't work (if the report was correct the module now didn't break with an oops but sound didn't work either). So I'm out of ideas, failed to see why stac92xx_mux_enum_info received uinfo == NULL (where uinfo comes from?) and passed it to snd_hda_input_mux_info. -- []'s Herton

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[alsa-devel] using ALSA OSS Emulation interface in long time will make voice delay
by Akio 28 Sep '07

28 Sep '07

Dear All: Is there any futher action to this topic "BF537Stamp Linphone audio falling behind". the URL is http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/? action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=22524 we have the same problem, but if we switch to alsa lib interface. the problem seems gone. Thanks in advance. Best Regards, Akio

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[alsa-devel] Asus M2NPV-VM (AD1986A) - Surround does work, snd-hda-intel dies on volume change
by Ma Begaj 28 Sep '07

28 Sep '07

Hi, Sorry if this email is not for devel mailing list, but this sounds to me like a bug or development problem. If I am wrong, I will post it on the user list. And for the long email ... I just wanted to describe everything I tried. I found a lot of messages about problems with sound chip AD1986A on motherboard Asus M2NPV-VM. It is almost always the same problem, surround does not work and sound mutes (module dies) when the user change volume. That is my problem too. I found a lot of solutions by using "model=3stack", sometimes with "position_fix=1", but that does not work for all the users. I checked the list, I read dozens of emails, but nothing helped. I found a few patches on the list, but all of them were for kernels < 2.6.22. I used Slackware 12.0 and model=3stack caused a terrible "weeping" sound on the rear left speaker, but the stereo worked. I read a lot of success stories with this motherboard and Ubuntu and after playing whole week long on Slackware (kernel 2.6.21.5 and 2.6.22.6), I decided to try it with Ubuntu Feisty (2.6.20 and 2.6.22.9), although I did not believe that Ubuntu should help. It actually helped partly because the weeping sound disappeared, but I had only stereo. When I changed the "Channel Mode" jack in alsamixer from 2ch to 6ch, sound muted and module died. The only way to get the sound back was to stop playing file, and rmmod/modprobe snd_hda_intel again. Using model=6stack gives me surround sound by using "speaker-test -c6 ..." (nice female voice guessed right every speaker) but module was still dying as soon as I changed volume. Does someone have an idea what to do with this chip ... I am really desperate and beginning to hate this great motherboard ... Thanks a lot. M.

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