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[alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 29 Apr '07

29 Apr '07

hi, I have an Acer Aspire 5100. I can play sound but with the build in mic, I can't record sound. The driver is the latest HG code. http://pastebin.ca/464406 [root@localhost ~]# lspci -nnv -d1002:437b 00:14.2 0403: 1002:437b (rev 01) Subsystem: 1025:009f Flags: bus master, slow devsel, latency 64, IRQ 19 Memory at c0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 [root@localhost ~]# vi /tmp/1 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/29/2007 04:18 AM, Cody Jung wrote: > Er...that appears to be a .diff file. Do I have to do something with it > for it to be workable, or...? Yes. You'd have to apply the patch (as it's called) to a recent kernel source tree, reconfigure the kernel adding this new VIVO driver to your config and then recompile and reinstall that kernel. Oh well, you posting to alsa-user and not alsa-devel should've been a hint. Rest assured the driver would've worked nicely :-/ There is another thing you can do. What was posted was just a wrapper around cs4231-lib in much the same way that the existing snd-cs4231 module is and that module you _will_ already have installed. That one doesn't do anything automatic though, so you will have to know how to manually enable your card first. First, open a root shell and look through the /sys/bus/pnp/devices directory for your VIVO. Under the "xx:yy.zz" sub-directories there you will find "id" files, and one of them will say "ENS1011". Example: # cat /sys/bus/pnp/devices/01\:01.00/id ENS1011 After having found it in there, activate the card: # echo activate >/sys/bus/pnp/devices/01\:01.00/resources and look at what resources were assigned to it by the PnP layer: # cat /sys/bus/pnp/devices/01\:01.00/resources state = active io 0x330-0x33f io 0x530-0x537 io 0x2100-0x213f irq 5 irq 9 dma 1 dma 0 You can now load the snd-cs4231 module specifying those card resources manually: # modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 Yes, 0x534. The second io resource + 4. This should've loaded without problem. Start up alsamixer: # alsamixer and up and unmute the PCM volume. With -> and <- you can walk through the sliders; ^ ups the one you're on, and "m" toggles the mute (the "MM" at the bottom should turn to a highlighted "OO"). PCM is the first slider. 90 seems to be an okay value. Quit alsamixer again with Alt-Q and now you should be able to hear things. If you have a plain "wav" file around somewhere, that makes for the quickest test: # aplay foo.wav No, I'm not going to tell you to plug in your speakers. If anything in the above doesn't work as expected, your system is not as expected and there's isn't much I can help with. If everything does work, you can automate it for the next time. I've attached (to try and avoid linewrap) the lines that you will need in /etc/modprobe.conf. Ubuntu has no doubt found a way to make that dark and muddy again so you might need to add it to an /etc/modprobe/whatever file instead. Please ask a fellow ubuntu victim for advice about translating this generic linux advice to your install. Once you do have that all setup, sound should "Just Work". You probably need to adjust the volume as specified above once only after which ubuntu will save it for the next boot, but if you don't hear anything ater rebooting, please check that first. Rene.

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/28/2007 07:06 AM, Cody Jung wrote: > As it suggests in the subject, I'm both a Linux-newbie and an > ALSA-newbie. I have an old clunker of a machine running Ubuntu Feisty > that I built out of parts left on curbs during spring cleanup, and that's > how I came across my sound card. It's an Ensoniq SoundscapeVivo (Not a > Vivo90, AFAIK). As far as I'm aware, there are no other VIVO's than the VIVO90. > I've gone through install and troubleshooting guides galore, but I can't > seem to get ALSA to recognize my card, and I'm confused and rather tired > of it all. > > Do you think you can help, and if so, what information do you need from > me? I probably can; I have both a Soundscape VIVO90 and a much older Soundscape S-2000. My VIVO90 is an ISA-PnP card and has a Ensoniq MARK5-RO, an OTTOR2C and a Analog Devices AD1845 codec: http://en.wikipedia.org/wiki/Ensoniq_Soundscape_VIVO90 I tested that when I acquired it a while ago and if I remember correctly, it just worked... mmm. come to think of it, it might be the case that I needed to add its PnP ID to the driver. Not sure anymore. Could you confirm whether or not your Soundscape is an ISA-PnP card and if so (that's also a way to find out if you don't know if it is) run "pnpdump" with it installed and look for a PnP-ID for the card in it? Something of the form "ENSxxxx". If your card is one of the older non-PnP soundscapes it's going to be a bit harder; they also need firmware and all that and I haven't tested that card myself yet. (please use the @alsa-project.org list addresses by the way; in the case of alsa-devel, the sourceforge one is gone even). Rene.

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[alsa-devel] Apple iMac pin configs
by Ivan N. Zlatev 28 Apr '07

28 Apr '07

Hello, I have a 20'' iMac machine with a hda-intel (SigmaTel 9221 A1 codec) and I have successfully been using the mac mini pin config, but there is one problem - the microphone doesn't work. The machine has a built-in microphone, built-in speakers, 1 digitial out jack and 1 for headphones. With the mac mini pin config I get 2 mics: "Front Mic" and "Mic" but none of those works. (alsamixer-macmini.gz for info). Today I spent some time on the search for the the correct pin configuration and it should be this [1] according to the windows drivers. This configuration causes amixer.gz, basically 1 "Mic" (which again doesn't record), but also "Mic as Output" and "Line in as output" (I don't have a line in as far as i know). I have also tried the macbook-pro and macbook introduced with 1.0.14rc3, but the mic doesn't work. I have attached * cat /proc/asound/card0/codec#0 * Mac OS X's complete dump of information of the audio driver - imac20-ioreg-audio.gz * amixer with the below config and the one with the macmini pin config Can you possible assist me to get the below pin config in tune, so that everything works? Regards. [1] ========== [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x0121E230 Action = hex: 01,0A,81,0C,81,10 [HKR\Settings\Pin\0B] CfgDflt = dword: 0x90A70120 [HKR\Settings\Pin\0C] CfgDflt = dword: 0x9017E110 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\0E] CfgDflt = dword: 0x400000FD Action = hex: 01,10,81,0A,81,0C [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0181E02E Action = hex: 09,0F [HKR\Settings\Pin\10] CfgDflt = dword: 0x1145E240 [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000FA [HKR\Settings\Pin\15] CfgDflt = dword: 0x400000FC [HKR\Settings\Pin\1B] CfgDflt = dword: 0x400000FB -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 27 Apr '07

27 Apr '07

Thanks to the help I've gotten on this list my alsa driver is working great. I got a little hung up not realizing that the alsa core basically provides everything you need to implement your own circular buffer if you need to use an intermediate buffer. The only question I still have is that if alsa allocates your DMA buffer for you based on snd_pcm_hardware_t buffer_bytes_max what exactly does snd_pcm_lib_preallocate_pages_for_all allocate buffers for? Does this allocate the buffers that would be passed to the copy callback? Thanks, Tim --- Tim Harvey <tim_harvey(a)yahoo.com> wrote: > Greetings, > > I'm setting out to write an ALSA driver for a very limited device thats > behind > a PLD. The actual codec is a uda1380 and does 16bit MSB stereo > encode/decode. > The way the device is driven the sampling rate is limited to 8KHz and data > transfer to/from the device I have to handle manually. > > I've read over the 'writing an ALSA driver' tutorial and have started the > driver but have run into a few questions: > > - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB > samples does the following snd_pcm_hardware_t look right?: > > static snd_pcm_hardware_t my_playback_hw = { > .info = (SNDRV_PCM_INFO_MMAP | > SNDRV_PCM_INFO_INTERLEAVED | > SNDRV_PCM_INFO_BLOCK_TRANSFER | > SNDRV_PCM_INFO_MMAP_VALID), > .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? > .rates = SNDRV_PCM_RATE_8000, // seems redundant? > .rate_min = 8000, > .rate_max = 8000, > .channels_min = 2, > .channels_max = 2, > .buffer_bytes_max = 32768, > .period_bytes_min = 4096, > .period_bytes_max = 32768, > .periods_min = 1, > .periods_max = 1024, > }; > > - I'm a little confused at how to deal with copying data from/to the device. > > It looks like I would implement the copy callback and copy the passed buffer > to > the device in the device specific manner I need to (playback), but I'm not > clear where to put data that I receive from the device (capture). I notice > that when the hw_params callback is called a dma buffer has been created by > ALSA in the substream->runtime struct. I assume this is created based on hte > buffer_bytes_max field above and I'm to copy data to this buffer? How do I > know where to copy it within the buffer and what data ALSA has already > consumed > from the buffer? > > - I might need to create an intermediate buffer between ALSA's rather large > buffers and my hardware buffer. Is there any driver that I could look at > that > does this? The tutorial talks about the vxpocket driver but that code looks > very foriegn from the rest of the ALSA drivers. > > - I'm not sure I understand what the 'period' is about. Its clear to me > that > I need to call snd_pcm_period_elapsed() periodically from either an IRQ or > other timed callback, but I'm not clear what the hardware pointer is or how > that converts to 'frames' and 'periods'. > > - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA > provides a mechanism for converting data from other formats? > > Thanks for any pointers, > > Tim >

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[alsa-devel] [ASoC] list corruption during AC97 registration
by Manuel Lauss 27 Apr '07

27 Apr '07

Hi Folks, I hit the following oops (on 2.6.21) when ASoC registers an AC97 codec. Happens both on MIPS and SH, for example the mips oops: asoc: AC97 HiFi <-> au1xpsc-ac97-1 mapping ok list_add corruption. prev->next should be next (80488f60), but was 865ffad8. (prev=865ffad8). <snip> Call Trace: [<80271290>] __list_add+0x68/0x94 [<802ac0e4>] bus_attach_device+0x48/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<c00882a8>] snd_soc_register_card+0x1d0/0x2c8 [snd_soc_core] [<c00b3268>] ac97_soc_probe+0x17c/0x1e4 [snd_soc_ac97] [<c0088f48>] soc_probe+0xb0/0x1ac [snd_soc_core] [<802ad410>] really_probe+0x148/0x268 [<802ac1cc>] bus_for_each_drv+0x50/0xa0 [<802ad758>] device_attach+0xe4/0x194 [<802ac0cc>] bus_attach_device+0x30/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<802aff24>] platform_device_add+0x13c/0x198 [<c00b71c8>] exm1200_mobo_ac97_init+0x1c8/0x1fc [snd_soc_exmmb_ac97] [<801478d0>] sys_init_module+0x1448/0x15b8 [<8010bcc0>] stack_done+0x20/0x3c System.map says: 80488e60 D ac97_bus_type As I said, it happens on both MIPS and SH (with machine and PCM drivers written by myself). With CONFIG_DEBUG_LIST disabled the code works and plays sound, but still I'd like to know what causes this list corruption. Thanks for any hints! -- Manuel Lauss

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[alsa-devel] dsnoop issues
by Gordon Miller 27 Apr '07

27 Apr '07

Hi All my snd_pcm_readi calls return too few frames. Also when I switch to stereo, the sound is muted. I cannot change this by adjusting period & buffer sizes. dsnoop is enabled by default. The documentation for snd_pcm_readi says that the only way that the frame count can be short is because of a signal or xrun. Well this clearly isn't the case. I think that direct snoop is the cause of this. Are there any remedies ? because I am roundly SICK of being told that my application must be wrong "because everyone else's works". I have spent the last 3 weeks debugging an app - that always worked properly - because of the faulty dsnoop implementation. It is abundantly clear that dsnoop is far from ready for use, but I am prepared to put some work into this. Whose idea was it to enable dsnoop by default, when it doesn't even work ? I was quite happy with OSS. Changing everything to be alsa entailed a lot of work, and I am beginning to think it was a complete waste of my time. ICE1712, 2.6.20.6 kernel, alsa 1.0.14rc1 Gordon

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[alsa-devel] New bugtracker categories
by Liam Girdwood 27 Apr '07

27 Apr '07

I'd like to request the following new categories for the ALSA bugtracker :- ASoC - platform - s3c24xx ASoC - platform - pxa2xx ASoC - platform - i.MX ASoC - platform - AT91xxxx ASoC - codecs ASoC - core Thanks Liam

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[alsa-devel] [ASoC] Sample format non available
by Markus Korber 27 Apr '07

27 Apr '07

Dear list, I'm currently implementing a ASoC solution for a custom chip and I've written a codec, I2S, PCM, and machine driver. However, when using aplay I get the following error (requested format was: 2): root:~# aplay -M -D hw:0,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:906: Sample format non available However, I've told each piece involved, that it is capable of playing S16_LE. So I don't know what I am missing here? Should I provide more debugging output like RULES_DEBUG output from sound/core/pcm_native.c? ,----[ CODEC DAI ] | #define CS4265_RATES (SNDRV_PCM_RATE_8000_192000) | #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) | | struct snd_soc_codec_dai cs4265_dai = { | .name = "CS4265", | .playback = { | .stream_name = "Playback", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | .capture = { | .stream_name = "Capture", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | [...] | }; `---- ,----[ CPU DAI ] | #define CHIP_I2S_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ | SNDRV_PCM_RATE_48000) | #define CHIP_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) | | struct snd_soc_cpu_dai chip_i2s_dai = { | /* DAI description */ | [...] | /* DAI callbacks */ | [...] | /* DAI capabilities */ | .playback = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | .capture = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | /* ops */ | [...] `---- ,----[ HW DAI ] | static const struct snd_pcm_hardware chip_pcm_hardware = { | .info = SNDRV_PCM_INFO_MMAP | | SNDRV_PCM_INFO_MMAP_VALID | /* For OSS emulation */ | SNDRV_PCM_INFO_BLOCK_TRANSFER | /* For OSS emulation */ | SNDRV_PCM_INFO_NONINTERLEAVED, | .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | | .channels_min = 2, /* Stereo */ | .channels_max = 2, | .period_bytes_min = 32, | .period_bytes_max = 8192, | .periods_min = 2, | .periods_max = 1024, /* PAGE_SIZE/sizeof(chip_dma_desc), */ | .buffer_bytes_max = 32 * 1024, | .fifo_size = 0, /* Still unused in ALSA? */ | }; `---- root:~# uname -a Linux 2.6.21 root:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). root:~# aplay --version aplay: version 1.0.14rc2 by Jaroslav Kysela <perex(a)suse.cz> root:~# aplay -M test.wav chip-pcm: Entered chip_pcm_open board_cs4265: Entered board_cs4265_startup Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_plug.c:840:(snd_pcm_plug_hw_refine_cchange) Unable to find an usable client format chip-pcm: Entered chip_pcm_hw_free chip-cs4265: Entered cs4265_shutdown chip-cs4265: Entered cs4265_write board_cs4265: Entered board_cs4265_shutdown chip-pcm: Entered chip_pcm_close ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IEC958_SUBFRAME_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: MU_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: A_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IMA_ADPCM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: GSM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: SPECIAL ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3BE aplay: set_params:878: Broken configuration for this PCM: no configurations available Regards, Markus Korber

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[alsa-devel] How to play a pcm file
by Flávio Yuiti 26 Apr '07

26 Apr '07

Greetings, I'm a new user of alsa api and I need to play pcm, adpcm and wav files using Alsa api. My code is below. I did a test with pcm file and the sound is playing very fast, the properties of the pcm file are 128kbps, 16 bits little endian, 1 channel, 8 khz, but I don't know how to set correctly the parameters in api. Please help me with this problem. Point me my errors. Thanks, Flavio int Sound::PlayFile(char *pszFileName, ULONG initialPoint, UINT qtdTimesToPlay, BOOL mix) { int dir; int result; int size; unsigned int val; snd_pcm_hw_params_t *params; snd_pcm_uframes_t frames; char* buffer; long tamanhoBuffer; FILE* file; int bytesLidos; // Open PCM device for playback. if((result = snd_pcm_open(&handleSound,"hw:0,0",SND_PCM_STREAM_PLAYBACK,0)) < 0) { fprintf(stderr,"\nUnable to open pcm device: %s",snd_strerror(result)); exit(1); } // Allocate a hardware parameters object. snd_pcm_hw_params_alloca(&hardwareParam); // Fill it in with default values. snd_pcm_hw_params_any(handleSound,hardwareParam); // Set the desired hardware parameters. // Interleaved mode. snd_pcm_hw_params_set_access(handleSound,hardwareParam,SND_PCM_ACCESS_RW_INTERLEAVED); // Signed 16-bit little-endian format. snd_pcm_hw_params_set_format(handleSound,hardwareParam,SND_PCM_FORMAT_S16_LE); // Two channels (stereo), one channel mono. snd_pcm_hw_params_set_channels(handleSound,hardwareParam,1); // Set rate. //snd_pcm_hw_params_set_rate(handleSound,hardwareParam,8000,0); // Set period size to 16384 frames. //snd_pcm_hw_params_set_period_size(handleSound,hardwareParam,16384,0); // Write the parameters to the driver. if((result = snd_pcm_hw_params(handleSound,hardwareParam)) < 0) { fprintf(stderr,"\nUnable to set hw parameters: %s",snd_strerror(result)); exit(1); } tamanhoBuffer = 16384; if((result = snd_pcm_prepare(handleSound)) < 0) { fprintf(stderr,"Cannot prepare audio interface for use: %s\n",snd_strerror(result)); exit(1); } file = fopen(pszFileName,"rb"); buffer = (char *)malloc(tamanhoBuffer); while((bytesLidos = fread(buffer,1,tamanhoBuffer,file)) > 0) { if((result = snd_pcm_writei(handleSound,buffer,tamanhoBuffer)) != tamanhoBuffer) { fprintf(stderr,"Write to audio interface failed: %s\n",snd_strerror(result)); exit(1); } } snd_pcm_drain(handleSound); snd_pcm_close(handleSound); free(buffer); fclose(file); return OK; } __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 2

[alsa-devel] [PATCH] PnP: Protocol for Media Vision Jazz16 sound cards
by Rask Ingemann Lambertsen 26 Apr '07

26 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add code to detect and PnP protocol to manage Media Vision Jazz16 sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20.6. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch depends on the patch "ALSA: Support Media Vision Jazz16 chips" because the patched include/sound/sb.h is needed. drivers/pnp/Kconfig | 12 drivers/pnp/Makefile | 1 drivers/pnp/jazz16pnp.c | 591 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 604 insertions(+) Main changes since the last patch: Request just one DMA channel (1, 3, 5 or 7). Request port 0x388-0x38b for the OPL3 alias to avoid conflicts. Use the recommended resources from the manual for the preferred option. More SB port bases: 0x210, 0x230 and 0x250. diff -rup linux-2.6.20.6-clean/drivers/pnp/Kconfig linux-2.6.20.6-ril/drivers/pnp/Kconfig --- linux-2.6.20.6-clean/drivers/pnp/Kconfig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/drivers/pnp/Kconfig 2007-04-07 20:10:35.000000000 +0200 @@ -37,5 +37,17 @@ source "drivers/pnp/pnpbios/Kconfig" source "drivers/pnp/pnpacpi/Kconfig" +# +# Jazz16 Plug and Play configuration +# +config JAZZ16PNP + bool "Jazz16 sound card Plug and Play support" + depends on PNP && ISA + help + Say Y here if you would like support for automatic configuration + of I/O ports, DMA channels and IRQ lines of Jazz16 sound cards. + + Say Y if you have such a card. Otherwise, say N. + endmenu diff -rup linux-2.6.20.6-clean/drivers/pnp/Makefile linux-2.6.20.6-ril/drivers/pnp/Makefile --- linux-2.6.20.6-clean/drivers/pnp/Makefile 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/drivers/pnp/Makefile 2007-04-07 20:10:35.000000000 +0200 @@ -7,3 +7,4 @@ obj-y := core.o card.o driver.o resourc obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPBIOS) += pnpbios/ obj-$(CONFIG_ISAPNP) += isapnp/ +obj-$(CONFIG_JAZZ16PNP) += jazz16pnp.o --- /dev/null 2007-04-11 19:30:17.320748640 +0200 +++ linux-2.6.20.6-ril/drivers/pnp/jazz16pnp.c 2007-04-11 21:41:39.000000000 +0200 @@ -0,0 +1,591 @@ +/* + * Media Vision Jazz16 Plug & Play support. + * + * Copyright (c) 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * This driver manages two devices: + * + * Device 0: The game port. + * Device 1: The SoundBlaster look-alike part. + * + * Sources: + * sound/oss/sb_common.c + * sys/dev/isa/sbdsp.c (OpenBSD) + */ + +#include <linux/types.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16 Plug & Play support"); +MODULE_LICENSE("GPL"); + +int jazz16pnp_disable = false; /* Disable Jazz16 PnP. */ +module_param_named(disable, jazz16pnp_disable, bool, false); +MODULE_PARM_DESC(disable, "Jazz16 Plug & Play disable switch"); + +int jazz16pnp_verbose = false; /* Verbosity level. */ +module_param_named(verbose, jazz16pnp_verbose, bool, false); +MODULE_PARM_DESC(verbose, "Jazz16 Plug & Play verbosity level"); + +#define PFX "jazz16pnp: " +#define PFX_DEBUG KERN_DEBUG PFX +#define PFX_INFO KERN_INFO PFX +#define PFX_WARN KERN_WARNING PFX + +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_JAZZ16_SET_DMAINTR 0xFB + +extern void *pnp_alloc(long size); +extern void pnp_free_option(struct pnp_option *option); + +static struct pnp_dev *jazz16_gameport_dev = NULL; +static struct pnp_dev *jazz16_sb_dev = NULL; +static struct resource *jazz16pnp_fm_res = NULL; + +static uint jazz16_sb_port = 0, jazz16_sb_irq = 0; +static uint jazz16_sb_dma1 = 0, jazz16_sb_dma2 = 0; +static uint jazz16_mpu401_port = 0, jazz16_mpu401_irq = 0; + +static uint jazz16pnp_count = 0; + +static struct pnp_protocol jazz16pnp_protocol; + +static bool __init jazz16pnp_easy_option(struct pnp_dev *dev, uint pri, + struct pnp_port *port, uint nports, + struct pnp_irq *irq, uint nirqs, + struct pnp_dma *dma, uint ndmas) +{ + struct pnp_option *opt; + struct pnp_port *new_port; + struct pnp_irq *new_irq; + struct pnp_dma *new_dma; + uint i; + + switch (pri) { + case PNP_RES_PRIORITY_PREFERRED: + case PNP_RES_PRIORITY_ACCEPTABLE: + case PNP_RES_PRIORITY_FUNCTIONAL: + opt = pnp_register_dependent_option(dev, pri); + break; + + default: + opt = pnp_register_independent_option(dev); + break; + } + if (!opt) + return false; + + for (i = 0; i < nports; i ++) { + new_port = pnp_alloc(sizeof(*new_port)); + if (!new_port) + goto out_no_mem; + memcpy(new_port, &port[i], sizeof(*new_port)); + pnp_register_port_resource(opt, new_port); + } + for (i = 0; i < nirqs; i ++) { + new_irq = pnp_alloc(sizeof(*new_irq)); + if (!new_irq) + goto out_no_mem; + memcpy(new_irq, &irq[i], sizeof(*new_irq)); + pnp_register_irq_resource(opt, new_irq); + } + for (i = 0; i < ndmas; i ++) { + new_dma = pnp_alloc(sizeof(*new_dma)); + if (!new_dma) + goto out_no_mem; + memcpy(new_dma, &dma[i], sizeof(*new_dma)); + pnp_register_dma_resource(opt, new_dma); + } + return true; + +out_no_mem: + return false; +} + +static struct pnp_port jazz16pnp_game_ports[] __initdata = { + { .min = SB_JAZZ16_CONFIG_PORT, + .max = SB_JAZZ16_CONFIG_PORT, + .size = 1, .align = 0, + .flags = PNP_PORT_FLAG_FIXED, }, +}; + +static int __init jazz16pnp_add_gameport(void) +{ + struct pnp_dev *dev; + struct pnp_id *dev_id; + + if (!(dev = pnp_alloc(sizeof(*dev)))) + goto out_no_dev; + if (!(dev_id = pnp_alloc(sizeof(*dev_id)))) + goto out_no_id; + if (!jazz16pnp_easy_option(dev, PNP_RES_PRIORITY_INVALID, + jazz16pnp_game_ports, 1, + NULL, 0, NULL, 0)) + goto out_no_option; + + dev->protocol = &jazz16pnp_protocol; + dev->capabilities = PNP_READ; + dev->active = 1; + dev->number = 0; + + pnp_init_resource_table(&dev->res); + dev->res.port_resource[0].start = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].flags = IORESOURCE_IO; + + memcpy(dev_id->id, "PNPb02f", PNP_ID_LEN); + pnp_add_id(dev_id, dev); + + jazz16_gameport_dev = dev; + pnp_add_device(dev); + return 0; + +out_no_option: + if (dev->independent) + pnp_free_option(dev->independent); + kfree(dev_id); +out_no_id: + kfree(dev); +out_no_dev: + return -ENOMEM; +}; + +static struct pnp_port jazz16pnp_pref_ports[] __initdata = { + { .min = 0x220, .max = 0x220, .align = 0x00, .size = 0x10 }, + { .min = 0x330, .max = 0x330, .align = 0x00, .size = 0x02 }, + { .min = 0x388, .max = 0x388, .align = 0x00, .size = 0x04 }, +}; + +static struct pnp_port jazz16pnp_ports[] __initdata = { + { .min = 0x210, .max = 0x260, .align = 0x10, .size = 0x10 }, + { .min = 0x300, .max = 0x330, .align = 0x10, .size = 0x02 }, + { .min = 0x388, .max = 0x388, .align = 0x00, .size = 0x04 }, +}; + +static struct pnp_irq jazz16pnp_pref_irqs[] __initdata = { + { .map[0] = (1 << 5), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 9), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_irq jazz16pnp_irqs[] __initdata = { + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_dma jazz16pnp_pref_dmas[] __initdata = { + { .map = (1 << 1), .flags = IORESOURCE_DMA_8BIT, }, +}; + +static struct pnp_dma jazz16pnp_dmas[] __initdata = { + { .map = (1 << 1) | (1 << 3) | (1 << 5) | (1 << 7), + .flags = IORESOURCE_DMA_8AND16BIT, }, +}; + +static int __init jazz16pnp_add_sb(void) +{ + struct pnp_dev *sbdev; + struct pnp_id *sbid; + + if (!(sbdev = pnp_alloc(sizeof(*sbdev)))) + goto out_no_sbdev; + if (!(sbid = pnp_alloc(sizeof(*sbid)))) + goto out_no_sbid; + + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_PREFERRED, + jazz16pnp_pref_ports, 3, + jazz16pnp_pref_irqs, 2, + jazz16pnp_pref_dmas, 1)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 3, + jazz16pnp_irqs, 2, + jazz16pnp_dmas, 1)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 3, + jazz16pnp_irqs, 1, + jazz16pnp_dmas, 1)) + goto out_no_sboption; + memcpy(sbid->id, "PNPb00f", PNP_ID_LEN); + pnp_add_id(sbid, sbdev); + + pnp_init_resource_table(&sbdev->res); + sbdev->capabilities = PNP_READ | PNP_WRITE | PNP_CONFIGURABLE | PNP_DISABLE; + sbdev->protocol = &jazz16pnp_protocol; + sbdev->number = 1; + + jazz16_sb_dev = sbdev; + pnp_add_device(sbdev); + return 0; + +out_no_sboption: + if (sbdev->dependent) + pnp_free_option(sbdev->dependent); + kfree(sbid); +out_no_sbid: + kfree(sbdev); +out_no_sbdev: + return -ENOMEM; +} + +/* Configure the SB and MPU ports. Passing 0 for sbport disables the Jazz16. */ +static void jazz16_port_setup(const uint sbport, const uint mpuport) +{ + u8 config; + + switch (sbport) { + case 0x210: + case 0x220: + case 0x230: + case 0x240: + case 0x250: + case 0x260: + config = sbport & 0x70; + jazz16_sb_port = sbport; + break; + + default: + config = 0; + jazz16_sb_port = 0; + break; + } + + switch (mpuport) { + case 0x300: + case 0x310: + case 0x320: + case 0x330: + config |= (mpuport >> 4) & 0x07; + jazz16_mpu401_port = mpuport; + break; + + default: + if (jazz16_sb_port) + BUG(); + } + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB port = %#x, MPU port = %#x.\n", + jazz16_sb_port, jazz16_mpu401_port); + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(config, SB_JAZZ16_CONFIG_PORT); +} + +static u8 jazz16_irq_config(uint irq) +{ + switch (irq) { + case 3: return 3; + case 5: return 1; + case 7: return 4; + case 9: return 2; + case 10: return 5; + case 15: return 6; + default: return 0; + } +} + +static u8 jazz16_dma_config(uint dma) +{ + switch (dma) { + case 1: return 1; + case 3: return 2; + case 5: return 3; + case 7: return 4; + default: return 0; + } +} + +/* The my_sbdsp_xxx functions were ripped from sound/isa/sb/sb_common.c. */ +#define BUSY_LOOPS 100000 + +static int my_sbdsp_command(unsigned int base, unsigned char val) +{ + int i; + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(base, STATUS)) & 0x80) == 0) { + outb(val, SBP1(base, COMMAND)); + return 1; + } + return 0; +} + +static int my_sbdsp_get_byte(unsigned int base) +{ + int val; + int i; + for (i = BUSY_LOOPS; i; i--) { + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + val = inb(SBP1(base, READ)); + return val; + } + } + return -ENODEV; +} + +static int my_sbdsp_reset(unsigned int base) +{ + int i; + + outb(1, SBP1(base, RESET)); + udelay(10); + outb(0, SBP1(base, RESET)); + udelay(30); + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + if (inb(SBP1(base, READ)) == 0xaa) + return 0; + else + break; + } + return -ENODEV; +} + +static int jazz16_irq_dma_setup(unsigned int base, uint sbirq, uint mpuirq, uint dma8, uint dma16) +{ + u8 dma_config, irq_config; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB irq %u, dma %u&%u, MPU irq %u.\n", + sbirq, dma8, dma16, mpuirq); + irq_config = jazz16_irq_config(sbirq) | jazz16_irq_config(mpuirq) << 4; + dma_config = jazz16_dma_config(dma8) | jazz16_dma_config(dma16) << 4; + if (!my_sbdsp_command(base, SB_JAZZ16_SET_DMAINTR)) + return 0; + + if (!my_sbdsp_command(base, dma_config)) + return 0; + jazz16_sb_dma1 = dma8; + jazz16_sb_dma2 = dma16; + + if (!my_sbdsp_command(base, irq_config)) + return 0; + jazz16_sb_irq = sbirq; + jazz16_mpu401_irq = mpuirq; + return 1; +} + +static int jazz16pnp_get_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + pnp_init_resource_table(res); + if (dev == jazz16_gameport_dev) { + res->port_resource[0].start = res->port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + res->port_resource[0].flags = IORESOURCE_IO; + } else if (dev == jazz16_sb_dev) { + if (!jazz16_sb_port) + return 0; + res->port_resource[0].start = jazz16_sb_port; + res->port_resource[0].end = jazz16_sb_port + 0x10 - 1; + res->port_resource[0].flags = IORESOURCE_IO; + + res->irq_resource[0].start = jazz16_sb_irq; + res->irq_resource[0].end = jazz16_sb_irq; + res->irq_resource[0].flags = IORESOURCE_IRQ; + if (jazz16_mpu401_irq) { + res->irq_resource[1].start = jazz16_mpu401_irq; + res->irq_resource[1].end = jazz16_mpu401_irq; + res->irq_resource[1].flags = IORESOURCE_IRQ; + } + + res->dma_resource[0].start = jazz16_sb_dma1; + res->dma_resource[0].end = jazz16_sb_dma1; + res->dma_resource[0].flags = IORESOURCE_DMA; + if (jazz16_sb_dma2 != jazz16_sb_dma1) { + res->dma_resource[1].start = jazz16_sb_dma2; + res->dma_resource[1].end = jazz16_sb_dma2; + res->dma_resource[1].flags = IORESOURCE_DMA; + } + res->port_resource[1].start = jazz16_mpu401_port; + res->port_resource[1].end = jazz16_mpu401_port + 2 - 1; + res->port_resource[1].flags = IORESOURCE_IO; + } else + return -EINVAL; + return 0; +} + +/* Disables the device if supported. */ +static int jazz16pnp_disable_resources(struct pnp_dev *dev) +{ + if (dev == jazz16_gameport_dev) { + return -ENOSYS; + } else if (dev == jazz16_sb_dev) { + if (jazz16_sb_port) + jazz16_irq_dma_setup(jazz16_sb_port, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + if (jazz16pnp_fm_res) { + release_resource(jazz16pnp_fm_res); + jazz16pnp_fm_res = NULL; + } + dev->active = false; + return 0; + } else { + return -EINVAL; + } +} + +#define PNP_RES_IS_SET(res,type,TYPE,i) \ + (((res)->type##_resource[i].flags & (IORESOURCE_##TYPE | IORESOURCE_UNSET)) \ + == IORESOURCE_##TYPE) + +static int jazz16pnp_set_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + unsigned int sbirq, mpuirq, dma1, dma2; + + if (dev == jazz16_gameport_dev) + return -ENOSYS; + + else if (dev == jazz16_sb_dev) { + if (PNP_RES_IS_SET(res, dma, DMA, 0)) + dma1 = res->dma_resource[0].start; + else + return -EINVAL; + if (PNP_RES_IS_SET(res, dma, DMA, 1)) + dma2 = res->dma_resource[1].start; + else + dma2 = dma1; /* "half duplex" operation. */ + if (PNP_RES_IS_SET(res, irq, IRQ, 0)) + sbirq = res->irq_resource[0].start; + else + return -EINVAL; + if (PNP_RES_IS_SET(res, irq, IRQ, 1)) + mpuirq = res->irq_resource[1].start; + else + mpuirq = 0; /* MPU-401 driver supports this. */ + if (! PNP_RES_IS_SET(res, port, IO, 0)) + return -EINVAL; + if (! PNP_RES_IS_SET(res, port, IO, 1)) + return -EINVAL; + + /* Block the OPL3 alias to avoid conflicts. */ + jazz16pnp_fm_res = request_region(0x388, 4, "SoundBlaster FM"); + /* Avoid setting the mpu port after setting the sb port + because doing so will glitch the mpu port from the + previous one to the new one. */ + jazz16_port_setup(res->port_resource[0].start, + res->port_resource[1].start); + jazz16_irq_dma_setup(jazz16_sb_port, sbirq, mpuirq, dma1, dma2); + dev->active = true; + return 0; + } else + return -EINVAL; +} +#undef PNP_RES_IS_SET + +static struct pnp_protocol jazz16pnp_protocol = { + .name = "Jazz16 PnP", + .get = jazz16pnp_get_resources, + .set = jazz16pnp_set_resources, + .disable = jazz16pnp_disable_resources, +}; + +static uint __init jazz16_check(unsigned int base) +{ + if (my_sbdsp_reset(base)) + return 0; + if (!(my_sbdsp_command(base, SB_DSP_GET_JAZZ_VERSION))) + return 0; + return SB_VERSION_IS_JAZZ16(my_sbdsp_get_byte(base)); +} + +static int __init jazz16pnp_init(void) +{ + int err = 0; + unsigned int base, mpubase; + + if (jazz16pnp_disable == 1) { + printk(PFX_INFO "Jazz16 Plug & Play support disabled.\n"); + return 0; + } + if (!request_region(SB_JAZZ16_CONFIG_PORT, 1, "Jazz16 PnP probe")) + return -EBUSY; + + for (mpubase = 0x300; mpubase <= 0x330; mpubase += 0x10) + if (request_region(mpubase, 2, "Jazz16 PnP probe")) + break; + if (mpubase > 0x330) + goto out_no_mpubase; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Scanning for Jazz16 cards...\n"); + + for (base = 0x260; base >= 0x210; base -= 0x10) { + if (!request_region(base, 16, "Jazz16 PnP probe")) + continue; + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Probing at %#x (with MPU at %#x).\n", + base, mpubase); + jazz16_port_setup(base, mpubase); + if (jazz16_check(base)) + jazz16pnp_count++; + jazz16_irq_dma_setup(base, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + release_region(base, 16); + if (jazz16pnp_count) + break; + } + if (base < 0x210) + goto out_no_sbbase; + if (!jazz16pnp_count) { + err = -ENODEV; + goto out_no_devices; + } + if (pnp_register_protocol(&jazz16pnp_protocol) < 0) + err = -ENOMEM; + else { + if (jazz16pnp_add_gameport()) + printk(PFX_WARN "Failed to add game port.\n"); + if (jazz16pnp_add_sb()) + printk(PFX_WARN "Failed to add SB port.\n"); + err = 0; + } +out_no_devices: + printk(PFX_INFO "%u Jazz16 card%s detected.\n", jazz16pnp_count, + jazz16pnp_count == 1 ? "" : "s"); +out_no_sbbase: + release_region(mpubase, 2); +out_no_mpubase: + release_region(SB_JAZZ16_CONFIG_PORT, 1); + return err; +} +module_init(jazz16pnp_init); + +static void __exit jazz16pnp_exit(void) +{ + pnp_unregister_protocol(&jazz16pnp_protocol); +} +module_exit(jazz16pnp_exit); -- Rask Ingemann Lambertsen

3 8

[alsa-devel] 2.6.21-rc7: known regressions with patches
by Adrian Bunk 26 Apr '07

26 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20 with patches available. 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 : 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 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> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available Subject : qla2xxx driver no longer works References : http://lkml.org/lkml/2007/4/16/49 http://lkml.org/lkml/2007/4/20/224 Submitter : David Miller <davem(a)davemloft.net> Robert Peterson <rpeterso(a)redhat.com> Handled-By : David Miller <davem(a)davemloft.net> Patch : http://lkml.org/lkml/2007/4/16/234 Status : patch was suggested

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[alsa-devel] playback open error:
by Ashlesha Shintre 25 Apr '07

25 Apr '07

Hi, I m trying to run the simple playback program that I have attached with this email -- and I get the following error: ALSA lib pcm_dmix.c:862:(snd_pcm_dmix_open) unable to open slave > Playback open error: Device or resource busy I am running this program as a function called wavplay (attached) -- however, if i run the same program, as part of the main function, then there is no problem -- What i mean is that if i basically just paste the code for the wavplay function in main and provide the input values (the pcm data to be played) -- then it can very well open the default device and play the pcm data that is being passed to it! Any pointers? Thanks, Ashlesha.

2 2

[alsa-devel] daily snapshot & version ID.
by Tobin Davis 25 Apr '07

25 Apr '07

Would it be possible to add a version ID update to the daily snapshots that you generate? It would make tracking a little easier, and we could get more testing on recent snapshots. That or spin another RC soon. -- Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] crash/reboot with rawmidi on ice1712 dual opteron
by Florian 25 Apr '07

25 Apr '07

Hi, on one particular machine, an IBM workstation (Intellistation A Pro with dual opteron 250, 4GB RAM), opening a rawmidi input port on an M-Audio 2496 card will cause an immediate reboot of the machine. I've tried with different SMP kernels and different ALSA versions (including 1.0.14rc3), but always the same. Syslog does not show anything. Using a USB MIDI interface works fine. Also, MIDI input on a similar workstation (Intellistation Z Pro, dual xeon, 1GB RAM) with M-Audio 2496 works without problems. Any ideas? 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!

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[alsa-devel] Digigram ISA cards
by Ash Willis 25 Apr '07

25 Apr '07

Hi, I'm thinking about trying to write a driver for some old Digigram cards soon, namely the PCX11 and PCX9. I might be wrong but it seems these cards natively support MPEG layers I and II and only actually support PCM by re-encoding in-driver. Does alsa-lib support MPEG conversions? Are there examples of drivers that already do this type of thing? Please forgive my ignorance. I grepped the sources and didn't find too much info. Ash =

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[alsa-devel] [1/3] 2.6.21-rc7: known regressions (v2)
by Adrian Bunk 25 Apr '07

25 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 : gammu no longer works References : http://lkml.org/lkml/2007/4/20/84 Submitter : Wolfgang Erig <Wolfgang.Erig(a)gmx.de> 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 : 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 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> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Wrong "From" email address selected earlier. Oops.

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

2 1

[alsa-devel] ICE 1724 MIDI support still broken?
by Daniel James 24 Apr '07

24 Apr '07

Hello ALSA team, We have a user with an ESI Juli@ whose Linux box hangs when any MIDI application starts up. I read that MIDI support is broken generally on ICE 1724 cards; is this still the case in recent driver releases? If so, is there any interest in fixing this long-standing issue? The user has indicated that his box would be available by ssh for any tests required. Cheers! Daniel

2 3

[alsa-devel] Standard config and plugin paths
by Takashi Iwai 24 Apr '07

24 Apr '07

Hi, while working on openSUSE 10.3 ALSA packaging, I had a discussion regarding the config and plugin paths. Basically, the current configs (typically in /usr/share/alsa or /etc/alsa) and binary plugins (/usr/lib/alsa-lib) are fairly strictly bound to the specific library, libasound.so.2. That is, if we publish a newer, incompatible version, libasound.so.3, these files will be very unlikely unusable, too. The suggestion was that we should keep these config and plugin files in the version-dependent paths so that no conflictions would occur at update. For example, instead of /usr/share/alsa, /usr/share/alsa2 or whatever. The fix would be easy -- just add some options to configure script of alsa-lib and alsa-plugins. So, the question is what path names are most suitable as the standard. For example, I find the following OK: - $DATADIR/alsa-1.0/ - $LIBDIR/alsa-lib-1.0/ Or, the deeper paths: - $DATADIR/alsa/1.0/ - $LIBDIR/alsa-lib/1.0/ For keeping the backward compatibility, we may implement fallback to read the old default path in addition, if the new path doesn't exist. In that case, the former suggestion would be cleaner. Any comments or suggestions? Takashi

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[alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 24 Apr '07

24 Apr '07

Greetings, I'm setting out to write an ALSA driver for a very limited device thats behind a PLD. The actual codec is a uda1380 and does 16bit MSB stereo encode/decode. The way the device is driven the sampling rate is limited to 8KHz and data transfer to/from the device I have to handle manually. I've read over the 'writing an ALSA driver' tutorial and have started the driver but have run into a few questions: - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB samples does the following snd_pcm_hardware_t look right?: static snd_pcm_hardware_t my_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? .rates = SNDRV_PCM_RATE_8000, // seems redundant? .rate_min = 8000, .rate_max = 8000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 32768, .period_bytes_min = 4096, .period_bytes_max = 32768, .periods_min = 1, .periods_max = 1024, }; - I'm a little confused at how to deal with copying data from/to the device. It looks like I would implement the copy callback and copy the passed buffer to the device in the device specific manner I need to (playback), but I'm not clear where to put data that I receive from the device (capture). I notice that when the hw_params callback is called a dma buffer has been created by ALSA in the substream->runtime struct. I assume this is created based on hte buffer_bytes_max field above and I'm to copy data to this buffer? How do I know where to copy it within the buffer and what data ALSA has already consumed from the buffer? - I might need to create an intermediate buffer between ALSA's rather large buffers and my hardware buffer. Is there any driver that I could look at that does this? The tutorial talks about the vxpocket driver but that code looks very foriegn from the rest of the ALSA drivers. - I'm not sure I understand what the 'period' is about. Its clear to me that I need to call snd_pcm_period_elapsed() periodically from either an IRQ or other timed callback, but I'm not clear what the hardware pointer is or how that converts to 'frames' and 'periods'. - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA provides a mechanism for converting data from other formats? Thanks for any pointers, Tim

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[alsa-devel] What does start_threshold mean, and what value should it contain?
by j t 24 Apr '07

24 Apr '07

Hi. I'm trying to debug a problem with xruns which happen when playing a wav file with vlc, but I can't reproduce the problems when using "aplay -M" to play the same wav file. (logged as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3031). I've checked the contents of /proc/asound/card0/pcm0p/sub0/ in both cases - they appear to be very similar except for the contents of sw_params: when I use vlc, start_threshold says 1, but when I use "aplay -M", start_threshold says 16384. So, what does start_threshold mean, is there a particular value that start_threshold "should" contain, and could this difference be the cause of the xruns? (The aplay man page doesn't seem to have any flags for controlling the value of this parameter, so it doesn't appear to be "user-configurable"). (Apologies if this is the wrong mailing list - I though that -devel would be better than -user, but I could be wrong). Thanks, Jaime

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Re: [alsa-devel] Fwd: No sound with ATI SB450
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 22:45:00 +1000, Ben DiDonc wrote: > > Tobin, Takashi, > Thank you very much for your help. > Takashi's last suggestion did it and I finally got a working ATI SB450 > sound card with Ubtuntu Feisty (kernel 2.6.20 and > alsa-driver-hg20070421). I did a (last) alsa-info.sh in case it helps > others, see http://pastebin.ca/454115 > Here's what it took: > modprobe snd_hda_intel probe_mask=8 model=auto > Without probe_mask, the ALC861 is not properly detected. Without > model=auto, playback doesnt work. > Let me know If I can be of any help or if you want me to try something else. As I requested, get /proc/asound/card0/codec#* files for both working and non-working states. Then we can compare what is not set up properly. probe_mask is a bit hard to fix right now. A least, we need to know which modem codec is harmful. Please check the codec#* file again with probe_mask=1. Takashi > Thanks again. > Ben > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 23 Apr 2007 21:58:17 +1000, > > Ben DiDonc wrote: > > > > > > Tobin, Takashi, > > > upon Takashi's advice, I tried loading hda-intel > > > (alsa-driver-hg20070421) using the probe_mask argument under 2.6.20. > > > With probe_mask=1, the codec isnt detected and only the modem shows up > > > (just like before). However, with probe_mask=8, the codec IS correctly > > > detected (alsa-info.sh at http://pastebin.ca/454059 ). syslog shows: > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 19 > > > ALSA .../alsa-driver-hg20070421/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:696: > > > codec_mask = 0x9 > > > ALSA .../alsa/alsa-driver-hg20070421/pci/hda/hda_codec.c:1819: > > > hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) > > > > > > However, there is still no playback (no muted control, all volumes at > > > 100% as shown by amixer in alsa-info.sh). Here again, capture works, I > > > successfully captured from the mic with arecord. > > > > Now you can try what I suggested, model=auto. And compare with > > codec#* files. If you have /proc/asound/card0/codec#* files on the > > working system, it'd be better, in addition... > > > > > > Takashi > > > > > > > > Can I draw your attention on something ? I m not sure about what i m > > > going to say (specially the part on NIDs) but i ll try. I know Tobin > > > already explained that this was normal, but if you have a look at the > > > post where I loaded edgy (2 posts ago), where the playback > > > works(2.6.17 with 1.0.12), you ll see that when I aplay something, > > > syslog shows only a single call to hda_codec_setup_stream with > > > NID=0x3. In all cases where playback doesnt work (1.0.14rcX under > > > 2.6.20), running aplay makes 4 calls to hda_codec_setup_stream with > > > NID=3, 6 ,5 and 4 in turn. Why does it try 6 5 and 4 ? Is it because 3 > > > fails ? We know that 3 should be used as its the one that works (under > > > 2.6.17). Now arent NIDs node IDs of codecs connected to the HDA > > > controller ? Is it possible that 6, 5 and 4 are mistakenly detected > > > and cannot be used for playback ? > > > Let me know if I m way off track. > > > > > Ben > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Mon, 23 Apr 2007 11:05:19 +0000, > > > > Ben DiDonc wrote: > > > > > > > > > > Takashi, > > > > > I did try passing model=auto to all 14 release candidates, > > > > > > > > Did you already try the HG version (or daily snapshot), right? > > > > > > > > > but without > > > > > success. the codec is never detected (see my previous posts for > > > > > alsa-info.sh outputs with both 2.6.20 and 2.6.17). > > > > > My problem is I cant get my sound card to work with kernel 2.6.20, > > > > > when it used to work with 2.6.17. > > > > > > > > The codec mask probed is 0x09, so you can pass either probe_mask=8 or > > > > probe_mask=1 to skip the modem detection (if it matters). > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > Ben > > > > > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > At Mon, 23 Apr 2007 10:50:18 +0000, > > > > > > Ben DiDonc wrote: > > > > > > > > > > > > > > Tobin, > > > > > > > Unfortunately, ACPI was on at all times. My laptop hardly ever boots > > > > > > > without ACPI (randomly, the SATA driver fails before it can even mount > > > > > > > the root partition). > > > > > > > I ran some more tests today: > > > > > > > I loaded Ubuntu Edgy from the live CD where the sound chip used to > > > > > > > work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works > > > > > > > straight away. From there , I ran the same series of test: download > > > > > > > and build (--with-debug=detect) the next version of alsa-driver, run a > > > > > > > capture from mic and try playing it back, check syslog and run > > > > > > > alsa-info.sh. I did this for successive versions of alsa-driver > > > > > > > (starting with 1.0.12) until it fails. > > > > > > > Here are the results: > > > > > > > > > > > > > > Alsa driver 1.0.12rc1(shipped with edgy): > > > > > > > capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 > > > > > > > > > > > > > > Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) > > > > > > > syslog lines after modprobe: > > > > > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: > > > > > > > codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: > > > > > > > line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 > > > > > > > (0x0/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, > > > > > > > dig_out=0x0, din_in=0x0 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, > > > > > > > fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 > > > > > > > > > > > > > > capture: working > > > > > > > //capture starting > > > > > > > hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, > > > > > > > channel=0, format=0x11 > > > > > > > //capture stopping > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > playback: working > > > > > > > //starting > > > > > > > ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, > > > > > > > fragsize=0x1000, format=0x11 > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, > > > > > > > channel=0, format=0x11 > > > > > > > //stopping > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) > > > > > > > modprobe snd-hda-intel: > > > > > > > hda_intel.c:680: codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, > > > > > > > cd=0x11, aux=0x0 > > > > > > > capture works fine, syslog output similar to 1.0.12 plus an extra > > > > > > > "hda-intel: Invalid position buffer, using LPIB read method instead" > > > > > > > in the middle of the capture. Playback works fine too, with the same > > > > > > > syslog entries as 1.0.12 > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) > > > > > > > modprobe: > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works (output similar to 1.0.12) > > > > > > > Playback does NOT work this time: > > > > > > > syslog shows a call to azx_pcm_prepare: and 4 calls to > > > > > > > hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out > > > > > > > my previous post) > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) > > > > > > > modprobe: the codec is correctly detected > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works. Playback does NOT work, same syslog entries as > > > > > > > 1.0.14rc1 with multiple calls to hda_codec_setup_stream > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) > > > > > > > modprobe: the codec isnt detected. > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > capture and playback dont work. > > > > > > > > > > > > > > Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy > > > > > > > (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but > > > > > > > not playback, 14rc3 doesnt detect the codec. > > > > > > > What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel > > > > > > > between releases 13 and 14 ... > > > > > > > > > > > > Try to pass model=auto to snd-hda-intel module, and check whether it > > > > > > works (make sure that you have unmuted/adjusted mixer elements before > > > > > > that). Then get /proc/asound/card0/codec#* files, and compare with > > > > > > the files without model option. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > > > >

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[alsa-devel] Compilation of latest alsa-hg
by dustin＠seznam.cz 24 Apr '07

24 Apr '07

Hello, please could you point me at a functioning description of latest complete hg tree compilation? The guides e.g. http://www.suseforums.net/index.php?showtopic=30156&pid=170707&st=0&#entry1… end up with /home/pavel/tmp/alsa-hg/alsa/alsa-driver/soc/codecs/../../alsa-kernel/soc/codecs/wm8753.c:1707: error: ‘I2C_DRIVERID_WM8753’ undeclared here (not in a function) Thanks a lot. Pavel.

2 1

Re: [alsa-devel] Functioning support for Prodigy 192
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 14:00:22 +0200 (CEST), dustin(a)seznam.cz wrote: > > Hello, > > Back here again :) > > While comparing media reading precision of various CD players I hit the following issues: > > 1. De-emphasis filter in AK4114 is turned on, distorting incoming data. A patch to disable: > > diff -r aed6a15c9398 pci/ice1712/prodigy192.c > --- a/pci/ice1712/prodigy192.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/prodigy192.c Mon Apr 23 12:47:01 2007 +0200 > @@ -714,7 +714,7 @@ static int prodigy192_ak4114_init(struct > { > static const unsigned char ak4114_init_vals[] = { > AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, > - AK4114_DIF_I24I2S, /* ice1724 expects I2S and provides clock */ > + AK4114_DIF_I24I2S | AK4114_DEM0 , /* ice1724 expects I2S and provides clock, DEM0 disables deemphasis filter */ > AK4114_TX1E, > AK4114_EFH_1024 | AK4114_DIT, /* default input RX0 */ > 0, Looks OK. > 2. All data recorded when master clock was provided by AK4114 (Multi Track Internal Clock= IEC958 Input) had double the sampling rate. For each input sample there were two identical output samples in the file. > > Here is where the FPGA on MIODIO gets introduced. Both MCKO1 and MCKO2 of ak4114 are fed to FPGA, which outputs master clock to SPMCLKIN of ice1724. Unfortunately my oscilloscope has just died, so experimentally I found out that only a combination of OCKS0=1, OCKS1 = 1 (128fs, 64fs output) and ice1724- VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct sampling rate. In order for this to work the FPGA must double the MCK01 rate (I cannot check the frequency now). > > It may actually make sense as it is probably the only way to support input sampling frequencies all the way up to 192kHz (128fs is max ak4114 can output at 192kHz) without having to switch the VT1724_MT_I2S_MCLK_128X bit in ice1724. In consumer mode the chips have no information about the sampling frequency (ak4114 does no detection) and the driver would not know when to set the bit. > > The problem is that due to switching sequence of Multi Track Internal Clock enum the VT1724_MT_I2S_MCLK_128X stays set (from the previous 192kHz option). I think it should not be set for ice1724 slave mode at all, here is the patch which fixes the sampling rate. Looks fine, too. Could you post a patch with a short patch summary and sign-off to apply to the upstream tree? thanks, Takashi > > Thanks, > > Pavel. > > diff -r aed6a15c9398 pci/ice1712/ice1724.c > --- a/pci/ice1712/ice1724.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/ice1724.c Mon Apr 23 13:57:19 2007 +0200 > @@ -1666,7 +1666,11 @@ static int snd_vt1724_pro_internal_clock > spin_lock_irq(&ice->reg_lock); > oval = inb(ICEMT1724(ice, RATE)); > if (ucontrol->value.enumerated.item[0] == spdif) { > + unsigned char i2s_oval; > outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); > + /* setting 256fs */ > + i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); > + outb( i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); > } else { > rate = rates[ucontrol->value.integer.value[0] % 15]; > if (rate <= get_max_rate(ice)) { > >

2 2

[alsa-devel] Buffer Overrun Messages
by John Dey 24 Apr '07

24 Apr '07

Hi, I have been driving fluidsynth with with aplaymidi and get the following: fluidsynth: ALSAsequencer buffer overrun, lost event. I have played around with buffer size and buffer number and also run with and without jackd. The sound out doesn't seem to be affect. Can anyone give me some guidance on how to eliminate the error. Could the error be caused by fluidsynth? I have also run the program on PS2/Yellowdog linux and CCRMA fc5 (x86) both with similar results. Thanks. John

2 1

[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 3

[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

Re: [alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, Unfortunately, ACPI was on at all times. My laptop hardly ever boots without ACPI (randomly, the SATA driver fails before it can even mount the root partition). I ran some more tests today: I loaded Ubuntu Edgy from the live CD where the sound chip used to work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works straight away. From there , I ran the same series of test: download and build (--with-debug=detect) the next version of alsa-driver, run a capture from mic and try playing it back, check syslog and run alsa-info.sh. I did this for successive versions of alsa-driver (starting with 1.0.12) until it fails. Here are the results: Alsa driver 1.0.12rc1(shipped with edgy): capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) syslog lines after modprobe: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, dig_out=0x0, din_in=0x0 ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture: working //capture starting hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 //capture stopping hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 playback: working //starting ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 //stopping ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) modprobe snd-hda-intel: hda_intel.c:680: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture works fine, syslog output similar to 1.0.12 plus an extra "hda-intel: Invalid position buffer, using LPIB read method instead" in the middle of the capture. Playback works fine too, with the same syslog entries as 1.0.12 alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) modprobe: ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works (output similar to 1.0.12) Playback does NOT work this time: syslog shows a call to azx_pcm_prepare: and 4 calls to hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out my previous post) alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) modprobe: the codec is correctly detected ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works. Playback does NOT work, same syslog entries as 1.0.14rc1 with multiple calls to hda_codec_setup_stream alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) modprobe: the codec isnt detected. ALSA hda_intel.c:696: codec_mask = 0x9 capture and playback dont work. Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but not playback, 14rc3 doesnt detect the codec. What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel between releases 13 and 14 ...

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[alsa-devel] ad1986a snd_hda_intel patch
by Dmitry Ursegov 23 Apr '07

23 Apr '07

Hello! I have an asus a8jn laptop and a integrated audio with Analog Devices AD1986A chip. $ lspci -vnn 0:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: ASUSTeK Computer Inc. Unknown device [1043:1447] Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> The audio seems to work properly on the right channel, but not on the left. All I get on the left channel is a loud screeching noise followed by a pop when the audio turns off. The microphon jack don't work also. I make a patch from the solution described in this http://mailman.alsa-project.org/pipermail/alsa-devel/2007-March/000093.html message: diff -Nur alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c --- alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c 2007-03-06 15:26:32.000000000 +0300 +++ alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c 2007-04-20 23:32:19.000000000 +0400 @@ -873,7 +873,7 @@ board_config = snd_hda_check_board_config(codec, AD1986A_MODELS, ad1986a_models, ad1986a_cfg_tbl); - switch (board_config) { +/* switch (board_config) { case AD1986A_3STACK: spec->num_mixers = 2; spec->mixers[1] = ad1986a_3st_mixers; @@ -912,6 +912,18 @@ spec->multiout.dig_out_nid = 0; break; } +*/ + + spec->num_mixers = 2; + spec->mixers[1] = ad1986a_3st_mixers; + spec->num_init_verbs = 3; + spec->init_verbs[1] = ad1986a_3st_init_verbs; + spec->init_verbs[2] = ad1986a_ch2_init; + spec->channel_mode = ad1986a_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; return 0; } I don't know, why snd_hda_check_board_config() return wrong value, but that workaround works. Thanks! Dmitry Ursegov

2 1

[alsa-devel] Strange code in
by Pete Zaitcev 23 Apr '07

23 Apr '07

Dear Takashi: Hans de Goede (on cc:) hits a problem when snd-usb-audio breaks with: "USB device is in the shutdown state, cannot create a card instance". This is how the place looks: chip = NULL; for (i = 0; i < SNDRV_CARDS; i++) { if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); goto __error; } chip = usb_chip[i]; break; } } The code looks broken to me, because dev at this time is a new device structure, freshly allocated. You cannot compare it with an old instance, and I don't know why you would want to. Unfortunately (for fortunately), I have zero clue about the way ALSA works, specifically rules for snd_device_ops.dev_free, so I am asking you to look at this problem. Yours, -- Pete

2 1

[alsa-devel] [PATCH] sound: fix incorrect use of platform_device_register()
by Akinobu Mita 23 Apr '07

23 Apr '07

The platform_device allocated by platform_device_alloc() should be added to the device hierarchy by platform_device_add() instead of platform_device_register(). Otherwise it will hit WARN_ON() in platform_device_register(). by illegal refcount. This patch fixes such incorrect usages in portman2x4 and mts64 drivers. Also it removes unnecessary trailing whitespaces. Cc: Jaroslav Kysela <perex(a)suse.cz> Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Index: 2.6-mm/sound/drivers/portman2x4.c =================================================================== --- 2.6-mm.orig/sound/drivers/portman2x4.c +++ 2.6-mm/sound/drivers/portman2x4.c @@ -676,13 +676,13 @@ static void __devinit snd_portman_attach struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; } Index: 2.6-mm/sound/drivers/mts64.c =================================================================== --- 2.6-mm.orig/sound/drivers/mts64.c +++ 2.6-mm/sound/drivers/mts64.c @@ -892,13 +892,13 @@ static void __devinit snd_mts64_attach(s struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; }

2 1

Re: [alsa-devel] Problem with Conexant
by Tobin Davis 23 Apr '07

23 Apr '07

Did you try adjusting the PCM volume levels? They may be set too low. That particular sound chip required a lot of manipulating to get functional, and is still very problematic. Unfortunately, I have no way of testing it, so users (like yourself) have to be my ears. The last time I worked with that particular configuration, the user that was testing said that it took a bit of tweaking with the PCM and other volume levels to get good sound. Tobin On Mon, 2007-04-23 at 08:11 +0307, Black X wrote: > after I install the last alsa-driver from the link you give me it > ( alsa-driver-hg20070421.tar.bz2 ) I still have the same problem. now > what I most to do to inform you about this issue? > > > Thank you. > > On Sun, 2007-04-22 at 19:27 -0700, Tobin Davis wrote: > > Could you try the latest alsa-driver snapshot from > > ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver? > > > > I can't tell what revision of Conexant code you are using based on > > your email. Anything after 20070413 should work better, and it will > > also be the most current. I haven't made any changes to the code > > since ~April 6. If you are still having audio issues, let me know. > > > > Tobin > > > > > > On Sun, 2007-04-22 at 23:31 +0307, Black X wrote: > > > Hello > > > > > > First at all I wanna thanks Tobin Davis for his support to get Conexant > > > audio devices working under Linux, and I was testing his last patchs, > > > but I had a little problems with my sound card, the first one is about > > > headphone, it is works but with noise, and when I'm maximize the volume > > > of headphone the sound quality starting so weak. > > > > > > The Second problem I had it is when I run the 3D Desktop Effects (XGL > > > with Compiz), that when I play any sound clip and moving any window or > > > rotate desktop cube that's making a noise with the sound, and the > > > quality of the sound clip it is so weak with slow playing in the sound. > > > > > > I make a little screen cast about this problem and you can see it at > > > this link: > > > http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html > > > > > > any way this is my alsa information: http://pastebin.ca/452453 > > > finaly my device information : > > > http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… > > > > > > I 'm sorry, but the English it's not my mother language. > > > > > > Thank you. > > > > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > > Tobin Davis <tdavis(a)dsl-only.net> > -- Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Ok, I try that before and try it now, still no change, and still have the same problem. When I highest the sound volume of headphone this makes sound starting noisy (and that's the same with PCM). About speaker it's work fine, but IF I moving any window or rotate the desktop cube this makes the sound very slow with noise.

1 0

[alsa-devel] partial fix for bug #2708: [ALSA - driver] PCI - hda-intel
by Jeffrey Arbuckle 23 Apr '07

23 Apr '07

Using alsa-driver-1.0.14rc3 with an iMac Core2 Duo 24" results in audio only available through the headphones. Additionally the sound is very low, even at the highest mixer setting. This problem is very similar to the bug recently fixed in 1.0.14rc2 with the audio on the MacPro. I was able to make a patch that applies the same fix for the MacPro to my iMac. The patch fixes the audio so that the built in speakers work normally, but now the headphones do not work at all. Obviously some more work is needed on this. The patch I've attached just adds my device's' PCI SSID to an existing switch statement. I am willing to provide some more testing if necessary. Here is my systems info: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Realtek Semiconductor Co., Ltd. Unknown device 0885 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 92400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 # cat /proc/asound/card0/codec#0 Codec: Realtek ALC885 Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x106b1000 Revision Id: 0x100103 # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3. Compiled on Apr 22 2007 for kernel 2.6.20-15-generic (SMP).

2 1

[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Hello First at all I wanna thanks Tobin Davis for his support to get Conexant audio devices working under Linux, and I was testing his last patchs, but I had a little problems with my sound card, the first one is about headphone, it is works but with noise, and when I'm maximize the volume of headphone the sound quality starting so weak. The Second problem I had it is when I run the 3D Desktop Effects (XGL with Compiz), that when I play any sound clip and moving any window or rotate desktop cube that's making a noise with the sound, and the quality of the sound clip it is so weak with slow playing in the sound. I make a little screen cast about this problem and you can see it at this link: http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html any way this is my alsa information: http://pastebin.ca/452453 finaly my device information : http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… I 'm sorry, but the English it's not my mother language. Thank you.

2 2

[alsa-devel] Qustions on dmix - aplay - hw_ptr - appl_ptr
by Matthew Foster 23 Apr '07

23 Apr '07

I am a driver developer that is new to ALSA. I have read the tutorial as an example on how to write an ALSA driver. I appreciate the documentation you have provided online. I was wondering if you might have time to possibly answer a question. I am having trouble understanding things in the memory/buffer management area. I am not writing a driver for a PCI device, I am writing a driver that acts as an interface from the ALSA to a render device. So I essentially have a circular buffer I am transferring the ALSA data to. I set up the ALSA buffer like this: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), ALSA_BUF_SIZE, ALSA_BUF_SIZE)) < 0); And later on in my driver after open is called I have a buffer polling loop that does the period notification and constantly monitors my render (ouput) buffer. I have my asound.conf setup like this: pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } This is to enable the dmix plugin so I can run aplay more than once and have the two streams mixed in. The problem I am having is that it seems that when I run aplay like this: "aplay close.wav" I only see this pointer moving: substream->runtime->status->hw_ptrbir And if I run aplay like this: "aplay -D plughw:0,0 close.wav" I only see this pointer moving: substream->runtime->control->appl_ptr This seems odd to me and currently in my code I just check which pointer moves and use that to calculate how much data I need to write to my render device. This is so I can support both use cases. Another major problem is that when I play a short wav file 25KB or so and use the dmix "aplay close.wav" it randomly doesn't play at sometimes. Also if I run "aplay close.wav" and then very quickly run "aplay close.wav" again at the same prompt I get corruption of the data which produces screeching. In the case where I run aplay one right after the other I notice that the hw_ptr moves differently (less) than if I run it, wait, run again. I am also noticing also that when I do the aplay over and over quickly that these values change (get lower) in my snd_pcm_prepare callback: chip_data->buf_size = snd_pcm_lib_buffer_bytes(substream); chip_data->period_buf_size = snd_pcm_lib_period_bytes(substream); It seems like those value should remain constant, and I use those values to calculate later in my polling thread to notify of a period lapse. It's almost like the runtime is telling me the period has changed and the buffer size has changed. But this only happens in this case with a short wav file and running aplay over and over quickly. Is there a way to make sure the hw_ptr doesn't get affected by multiple instances of aplay trying to render out small clips. It seems like maybe I am setting up something incorrectly or maybe not blocking/ waiting correctly. I am at a loss in trying to figure this out, wondering if you would be so kind as to comment on my situation. I really appreciate your time in reading this email. Matt

2 2

[alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, I downloaded and installed alsa-driver, alsa-lib and alsa-utils, but still no sound coming out of those speakers... I ran alsa-info.sh and the results are at http://pastebin.ca/451128 I am clueless as to what to do next, but I m ready to try anything. Any help would be greatly appreciated. Ben On 4/21/07, Tobin Davis < tdavis(a)dsl-only.net> wrote: > > Try downloading a current daily snapshot from ftp://ftp.suse.com/pub/projects/alsa/snapshot/ > . > > There was an issue with the ATI HD Audio detection that was fixed after > 14rc3 was released. > > Tobin > > On Sat, 2007-04-21 at 16:00 +1000, Ben DiDonc wrote: > > The output of alsa-info.sh is available at http://pastebin.ca/451007 > > > On 4/21/07, *Tobin Davis* < tdavis(a)dsl-only.net> wrote: > > Could you download an run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > , then post the resulting link? This will help determine what setup you > have. > > Tobin Davis > > > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. > Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I > just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) > and the only mixer settings available are those of the modem, named off-hook > and caller-id (see output of amixer, aplay -l , and the rest in attached > file output_feisty) I decided to upgrade alsa-driver and alsa-lib to > 1.0.14rc2. after downloading the package, I unloaded all sound related > modules (everything with "snd_" in it) in the right order and did: > ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; > sudo modprobe snd_hda_intel A minor improvement: the mixer now shows > controls for both the modem and the sound card (see > output_after_upgrade_to_rc2), but unfortunately, there is still no sound > coming out of those speakers (even though alsamixer shows control for PCM > and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it > a shot. This time again, only modem related settings appeared in alsamixer, > and no sound card stuff (see aplay and amixer in > output_after_upgrade_to_rc3), and no sound either. Does anyone know how to > get some sound out of these speakers (given that it used to work with > alsa1.0.12rc1...) ? Thanks for your help Ben > > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> > > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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Re: [alsa-devel] iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 22 Apr '07

22 Apr '07

[Nice analysis cut out..] At Tue, 03 Apr 2007 10:17:16 +0200, Dominique Dumont wrote: > > If the stream format is not valid when the digi converter is set up, > there's a chance that the digi converter goes belly up. > > I reckon that setting the stream *then* the digi converter would be safer. OK, now question is whether we need to turn off DIGI_CONV* once before setting it up again. Anyway, a test patch is below. It forces to set DIGI_CONV* verbs in prepare. Give it a try. thanks, Takashi diff -r 05ecca0fba92 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_codec.c Tue Apr 03 15:28:47 2007 +0200 @@ -1083,6 +1083,19 @@ static int snd_hda_spdif_default_put(str return change; } +static void setup_digi_convert(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned short val = codec->spdif_ctls; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | + AC_AMP_SET_OUTPUT | + ((val & 1) ? 0 : 0x80)); +} + static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1114,10 +1127,7 @@ static int snd_hda_spdif_out_switch_put( change = codec->spdif_ctls != val; if (change || codec->in_resume) { codec->spdif_ctls = val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | - AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); + setup_digi_convert(codec, nid); } mutex_unlock(&codec->spdif_mutex); return change; @@ -1901,6 +1911,21 @@ int snd_hda_multi_out_dig_open(struct hd return 0; } +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + mutex_lock(&codec->spdif_mutex); + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} + /* * release the digital out */ @@ -1949,6 +1974,8 @@ int snd_hda_multi_out_analog_prepare(str mout->dig_out_used = 0; snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); } + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); } mutex_unlock(&codec->spdif_mutex); diff -r 05ecca0fba92 pci/hda/hda_local.h --- a/pci/hda/hda_local.h Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_local.h Tue Apr 03 15:31:29 2007 +0200 @@ -148,6 +148,11 @@ struct hda_multi_out { int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout); +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, diff -r 05ecca0fba92 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/patch_realtek.c Tue Apr 03 15:32:23 2007 +0200 @@ -1916,6 +1916,17 @@ static int alc880_dig_playback_pcm_open( return snd_hda_multi_out_dig_open(codec, &spec->multiout); } +static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1984,7 +1995,8 @@ static struct hda_pcm_stream alc880_pcm_ /* NID is set in alc_build_pcms */ .ops = { .open = alc880_dig_playback_pcm_open, - .close = alc880_dig_playback_pcm_close + .close = alc880_dig_playback_pcm_close, + .prepare = alc880_dig_playback_pcm_prepare }, };

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[alsa-devel] Please ignore, test.
by Rene Herman 22 Apr '07

22 Apr '07

Body Attachment

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Re: [alsa-devel] No sound with ATI SB450
by Ben DiDonc 21 Apr '07

21 Apr '07

The output of alsa-info.sh is available at http://pastebin.ca/451007 On 4/21/07, Tobin Davis <tdavis(a)dsl-only.net> wrote: > > Could you download an run > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh , then post the > resulting link? This will help determine what setup you have. > > Tobin Davis > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > HiI have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450.Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy).I just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa1.0.14rc1) and the only mixer settings available are those of themodem, named off-hook and caller-id (see output of amixer, aplay -l ,and the rest in attached file output_feisty) > I decided to upgrade alsa-driver and alsa-lib to 1.0.14rc2. afterdownloading the package, I unloaded all sound related modules(everything with "snd_" in it) in the right order and did:./configure --with-oss=yes --with-cards=hda-intel;make; sudo makeinstall; sudo modprobe snd_hda_intelA minor improvement: the mixer now shows controls for both the modemand the sound card (see output_after_upgrade_to_rc2), butunfortunately, there is still no sound coming out of those speakers(even though alsamixer shows control for PCM and the rest...). > Finally, I downloaded alsa-driver-1.0.14rc3 and gave it a shot. Thistime again, only modem related settings appeared in alsamixer, and nosound card stuff (see aplay and amixer inoutput_after_upgrade_to_rc3), and no sound either. > Does anyone know how to get some sound out of these speakers (giventhat it used to work with alsa1.0.12rc1...) ? > Thanks for your helpBen > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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[alsa-devel] No sound with ATI SB450
by Ben DiDonc 21 Apr '07

21 Apr '07

Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) and the only mixer settings available are those of the modem, named off-hook and caller-id (see output of amixer, aplay -l , and the rest in attached file output_feisty) I decided to upgrade alsa-driver and alsa-lib to 1.0.14rc2. after downloading the package, I unloaded all sound related modules (everything with "snd_" in it) in the right order and did: ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; sudo modprobe snd_hda_intel A minor improvement: the mixer now shows controls for both the modem and the sound card (see output_after_upgrade_to_rc2), but unfortunately, there is still no sound coming out of those speakers (even though alsamixer shows control for PCM and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it a shot. This time again, only modem related settings appeared in alsamixer, and no sound card stuff (see aplay and amixer in output_after_upgrade_to_rc3), and no sound either. Does anyone know how to get some sound out of these speakers (given that it used to work with alsa1.0.12rc1...) ? Thanks for your help Ben

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[alsa-devel] [PATCHv2] ALSA: Support Media Vision Jazz16 chips
by Rask Ingemann Lambertsen 20 Apr '07

20 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add support for Jazz16 features to the common SoundBlaster code. - Add a front end for Jazz16 based sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20.6. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch is a prerequisite for the Jazz PnP patch, which needs the patched include/sound/sb.h. Documentation/sound/alsa/ALSA-Configuration.txt | 10 include/sound/sb.h | 5 sound/isa/Kconfig | 15 + sound/isa/sb/Makefile | 2 sound/isa/sb/jazz16.c | 183 ++++++++++++++ sound/isa/sb/sb8_main.c | 98 +++++++ sound/isa/sb/sb_common.c | 12 sound/isa/sb/sb_mixer.c | 3 8 files changed, 323 insertions(+), 5 deletions(-) Changes since the last patch: The same DMA channel is now used for both 8-bit and 16-bit sound. DSP high-speed mode isn't used at all now for the Jazz16. A 16-bit DMA buffer can now be 128 kB instead of 64 kB. diff -rup linux-2.6.20.6-clean/Documentation/sound/alsa/ALSA-Configuration.txt linux-2.6.20.6-ril/Documentation/sound/alsa/ALSA-Configuration.txt --- linux-2.6.20.6-clean/Documentation/sound/alsa/ALSA-Configuration.txt 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/Documentation/sound/alsa/ALSA-Configuration.txt 2007-04-07 17:31:38.000000000 +0200 @@ -1120,6 +1120,16 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based on Media Vision Jazz16 chip (PnP only) + + This module supports multiple cards and PnP. + + Note: You will most likely want to enable Jazz16 PnP support + (CONFIG_JAZZ16PNP) in the kernel. + Module snd-korg1212 ------------------- diff -rup linux-2.6.20.6-clean/include/sound/sb.h linux-2.6.20.6-ril/include/sound/sb.h --- linux-2.6.20.6-clean/include/sound/sb.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/include/sound/sb.h 2007-04-07 17:31:38.000000000 +0200 @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ @@ -142,6 +143,7 @@ struct snd_sb { #define SB_DSP_HI_INPUT_AUTO 0x98 #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +267,9 @@ struct snd_sb { #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ diff -rup linux-2.6.20.6-clean/sound/isa/Kconfig linux-2.6.20.6-ril/sound/isa/Kconfig --- linux-2.6.20.6-clean/sound/isa/Kconfig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/Kconfig 2007-04-07 17:31:38.000000000 +0200 @@ -248,6 +248,21 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Media Vision Jazz16 based + soundcards. + + You probably want to enable Jazz16 Plug and Play support + (CONFIG_JAZZ16PNP) also. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND diff -rup linux-2.6.20.6-clean/sound/isa/sb/Makefile linux-2.6.20.6-ril/sound/isa/sb/Makefile --- linux-2.6.20.6-clean/sound/isa/sb/Makefile 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/Makefile 2007-04-07 17:31:38.000000000 +0200 @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb8_main.c linux-2.6.20.6-ril/sound/isa/sb/sb8_main.c --- linux-2.6.20.6-clean/sound/isa/sb/sb8_main.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb8_main.c 2007-04-12 00:03:35.000000000 +0200 @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Wed Apr 11 23:56:18 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements (mostly ported from sound/oss/sb_common.c). */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,13 +113,52 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + if (c->max > 1) { + unsigned int num = 0, den = 0; + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char stereo, format; + stereo = runtime->channels > 1; rate = runtime->rate; switch (chip->hardware) { case SB_HW_PRO: @@ -123,6 +174,7 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -134,8 +186,14 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + else + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -167,6 +225,8 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->dma8 > 3) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); @@ -233,7 +293,9 @@ static int snd_sb8_capture_prepare(struc struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char stereo, format; + stereo = runtime->channels > 1; rate = runtime->rate; switch (chip->hardware) { case SB_HW_PRO: @@ -250,6 +312,7 @@ static int snd_sb8_capture_prepare(struc break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -261,10 +324,14 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + else + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (stereo || chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -279,6 +346,8 @@ static int snd_sb8_capture_prepare(struc snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->dma8 > 3) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); @@ -443,6 +512,19 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -465,6 +547,14 @@ static int snd_sb8_open(struct snd_pcm_s } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -512,6 +602,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc; if (rpcm) *rpcm = NULL; @@ -524,9 +615,10 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + max_prealloc = (chip->dma8 > 3) ? 128 * 1024 : 64 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb_common.c linux-2.6.20.6-ril/sound/isa/sb/sb_common.c --- linux-2.6.20.6-clean/sound/isa/sb/sb_common.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb_common.c 2007-04-11 20:23:55.000000000 +0200 @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb_mixer.c linux-2.6.20.6-ril/sound/isa/sb/sb_mixer.c --- linux-2.6.20.6-clean/sound/isa/sb/sb_mixer.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb_mixer.c 2007-04-07 17:31:38.000000000 +0200 @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: --- /dev/null 2007-04-11 19:30:17.320748640 +0200 +++ linux-2.6.20.6-ril/sound/isa/sb/jazz16.c 2007-04-05 23:30:27.000000000 +0200 @@ -0,0 +1,183 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * There is also an MPU-401 interface. + * + * AFAIK, the following systems have an on-board Jazz16 chip: + * IBM PPS Model 6015 + * Texas Intstruments/Acer Travelmate TM5000 series. + * Please tell me if this driver works with these boards or not. + */ + +#include <sound/driver.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Jazz16 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Jazz16 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Jazz16 soundcard."); + +#define PFX "jazz16: " +#define PFX_WARN KERN_WARNING PFX +#define PFX_ERR KERN_ERR PFX + +static struct pnp_driver snd_jazz16_pnp_driver; + +static irqreturn_t jazz16_interrupt(int irq, void *card) +{ + return snd_sb8dsp_interrupt(card); +} + +static struct pnp_device_id snd_jazz16_pnpids[] = { + { .id = "PNPb00f" }, + { .id = "" } +}; +MODULE_DEVICE_TABLE(pnp, snd_jazz16_pnpids); + +static int __devinit snd_jazz16_pnp_probe(struct pnp_dev *pnp_dev, + const struct pnp_device_id *pnp_id) +{ struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int err; + uint sbport, sbirq, sbdma, mpuport, mpuirq; + static uint dev_num = 0; + + if (enable[dev_num]) + card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); + else + card = NULL; + dev_num++; + if (card == NULL) + return enable[dev_num] ? -ENOMEM : -ENODEV; + snd_card_set_dev(card, &pnp_dev->dev); + pnp_set_drvdata(pnp_dev, card); + /* TODO use pnp_port_valid(), pnp_port_flags(), pnp_port_length()... */ + sbport = pnp_port_start(pnp_dev, 0); + sbirq = pnp_irq(pnp_dev, 0); + sbdma = pnp_dma(pnp_dev, 0); + err = snd_sbdsp_create(card, sbport, sbirq, jazz16_interrupt, + sbdma, -1, SB_HW_AUTO, &chip); + if (err < 0) + goto out_err; + + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at %#x, irq %u, dma %u", + chip->name, sbport, sbirq, chip->dma8); + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(PFX_ERR "Not a Jazz16 chip at %#x.\n", sbport); + snd_card_free(card); + return -ENODEV; + } + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto out_err; + + err = snd_sbmixer_new(chip); + if (err < 0) + goto out_err; + + err = snd_opl3_create(card, sbport, sbport + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(PFX_WARN "No OPL device found, skipping.\n"); + else { + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto out_err; + } + if (pnp_port_valid(pnp_dev, 1) + && !(pnp_port_flags(pnp_dev, 1) & IORESOURCE_DISABLED)) { + int mpuirqflags; + if (!pnp_irq_valid(pnp_dev, 1) + || pnp_irq_flags(pnp_dev, 1) & IORESOURCE_DISABLED) { + mpuirq = -1; + mpuirqflags = 0; + } else { + mpuirq = pnp_irq(pnp_dev, 1); + mpuirqflags = IRQF_DISABLED; + } + mpuport = pnp_port_start(pnp_dev, 1); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport, 0, mpuirq, + mpuirqflags, NULL); + if (err < 0) + snd_printk(PFX_WARN "MPU-401 device not configured, skipping.\n"); + } + err = snd_card_register(card); + if (err >= 0) + return 0; +out_err: + snd_card_free(card); + return err; +} + +static void __devexit snd_jazz16_pnp_remove(struct pnp_dev *dev) +{ + snd_card_free(pnp_get_drvdata(dev)); + pnp_set_drvdata(dev, NULL); +} + +static struct pnp_driver snd_jazz16_pnp_driver = { + .name = "Jazz16", + .id_table = snd_jazz16_pnpids, + .probe = snd_jazz16_pnp_probe, + .remove = __devexit_p(snd_jazz16_pnp_remove), +}; + +static int __devinit alsa_card_jazz16_init(void) +{ + return pnp_register_driver(&snd_jazz16_pnp_driver); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + pnp_unregister_driver(&snd_jazz16_pnp_driver); +} + +module_init(alsa_card_jazz16_init); +module_exit(alsa_card_jazz16_exit); -- Rask Ingemann Lambertsen

3 3

[alsa-devel] [Alsa-devel] Read the input of a soundcard with ALSA
by Nyx 20 Apr '07

20 Apr '07

Hi, I have some problems to read the input of my soundcard using Alsa and I don't really understand how to have access to the input of my soundcard. I follow : http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht… # So, in my program, I have created two handles for capture and playback : char *device = "hw:0,0"; snd_output_t *output; snd_input_t *input; snd_pcm_t *phandle, *chandle; # then I connect the phandle to the out err = snd_output_stdio_attach(&output, stdout, 0); if (..) # Do I have to connect the chandle with the input ? # err = snd_input_stdio_attach(&input, stdin, 0); # I open the access to PCM err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) if (..) err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) if(..) # Hardware and software Parameters ## Hardware Parameters snd_pcm_hw_params; int err; err = snd_pcm_hw_params_any(phandle, params); if(..); err = snd_pcm_hw_params_set_access(phandle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); if(..); err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); if(..); err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), if(..); err = snd_pcm_hw_params(phandle, params); ##Software parameters err = snd_pcm_sw_params(phandle, params); if(..); err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); if(..); err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); # same configuration for phandle # Link output to input and start err = snd_pcm_link(chandle, phandle); if(..); err = snd_pcm_start(chandle); if(..); #Then I want to read my input frames_in = 0; in_max = 0; latency = 28; buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); while (1) { r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); } # description of the function long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t *max) { long r; do { r = snd_pcm_readi(handle, buf, len); } while (r == -EAGAIN); if (r > 0) { *frames += r; if ((long)*max < r) *max = r; } printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, buf[i]); return r; } So I compile it and I run it, and I have $ read = Broken Pipe, len = 28 and buf[..] = 0.000 Anyone can help me to read the input of my sound card ? Thank you -- K

1 0

Re: [alsa-devel] [PATCH 16/16] alsa sound support for mips
by Ralf Baechle 20 Apr '07

20 Apr '07

On Sun, Apr 15, 2007 at 11:26:05PM +0800, tiansm(a)lemote.com wrote: (Adding a few more people to the cc'list) > sound/core/pcm_native.c | 10 ++++++++++ > sound/core/sgbuf.c | 9 +++++++++ > 2 files changed, 19 insertions(+), 0 deletions(-) > > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 3e276fc..9005bac 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -3145,7 +3145,11 @@ static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, > return NOPAGE_OOM; /* XXX: is this really due to OOM? */ > } else { > vaddr = runtime->dma_area + offset; > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) Please use CONFIG_MIPS instead of __mips__ in #if / #ifdefs. The question if #ifdefing is the right approach to solve this problem is something else but I think no, .... > + page = virt_to_page(CAC_ADDR(vaddr)); > +#else > page = virt_to_page(vaddr); > +#endif So this is needed because the MIPS virt_to_page is returning a unsuitable value if vaddress is not a KSEG0 (64-bit: cached XKPHYS) address which is what GFP allocations and the slab will return. So now we have to deciede if a) the MIPS __pa() should be changed to handle uncached addresses. b) the sound code here is simply broken. Some drivers seem to allocate runtime->dma_area from vmalloc, so this whole area in the sound code is looking like built on quicksand ... > } > get_page(page); > if (type) > @@ -3261,6 +3265,12 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) > substream = pcm_file->substream; > snd_assert(substream != NULL, return -ENXIO); > > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) > + /* all mmap using uncached mode */ > + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); > + area->vm_flags |= ( VM_RESERVED | VM_IO); VM_RESERVED will prevent the buffer from being freed. I assume that is another workaround for some kernel subsystem blowing up when being fed a pointer to an uncached RAM address? This smells like a memory leak. > +#endif > + > offset = area->vm_pgoff << PAGE_SHIFT; > switch (offset) { > case SNDRV_PCM_MMAP_OFFSET_STATUS: > diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c > index cefd228..535f0bc 100644 > --- a/sound/core/sgbuf.c > +++ b/sound/core/sgbuf.c > @@ -91,12 +91,21 @@ void *snd_malloc_sgbuf_pages(struct device *device, > } > sgbuf->table[i].buf = tmpb.area; > sgbuf->table[i].addr = tmpb.addr; > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) > + sgbuf->page_table[i] = virt_to_page(CAC_ADDR(tmpb.area)); VM_RESERVED will prevent the buffer from being freed. I assume that is another workaround for some kernel subsystem blowing up when being fed a pointer to an uncached RAM address? This smells like a memory leak. > +#else > sgbuf->page_table[i] = virt_to_page(tmpb.area); > +#endif > sgbuf->pages++; > } > > sgbuf->size = size; > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) > + /* maybe we should use uncached accelerated mode */ > + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP | VM_IO, pgprot_noncached(PAGE_KERNEL)); > +#else > dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); > +#endif I would suggest to get rid of this ifdef with a new arch-specific function like vmap_io_buffer which will do whatever a platform seems fit for this case? > if (! dmab->area) > goto _failed; > return dmab->area; Ralf

3 2

[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by Pablo N. Hess 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

2 1

[alsa-devel] [KJ][PATCH] sound: SPIN_LOCK_UNLOCKED cleanup
by Milind Arun Choudhary 19 Apr '07

19 Apr '07

SPIN_LOCK_UNLOCKED cleanup,use __SPIN_LOCK_UNLOCKED instead Signed-off-by: Milind Arun Choudhary <milindchoudhary(a)gmail.com> --- soc/at91/at91-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c index 9fc0c03..d65a47d 100644 --- a/sound/soc/at91/at91-i2s.c +++ b/sound/soc/at91/at91-i2s.c @@ -151,20 +151,20 @@ static struct at91_ssc_info { } ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = 0, .initialized = 0, }, #if NUM_SSC_DEVICES == 3 { .name = "ssc1", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = 0, .initialized = 0, }, { .name = "ssc2", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = 0, .initialized = 0, }, -- Milind Arun Choudhary

1 0

[alsa-devel] CS4270 supported?
by Timur Tabi 19 Apr '07

19 Apr '07

I've never worked with ALSA before, but I'm about to start developing on an ALSA driver for our hardware, which has a CS4270 on it attached via an I2S interface. After scanning the source code and mailing lists, I see no mention of the CS4270. So does that mean I would need to write an ASoC codec driver for the CS4270? If so, does the data sheet (http://www.cirrus.com/en/pubs/proDatasheet/CS4270_PP1.pdf) contain enough information? -- Timur Tabi Linux Kernel Developer @ Freescale

4 7

[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by natunobilis＠gentoobr.org 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

1 0

[alsa-devel] Surround problems on ALC883
by Pavel Patz 18 Apr '07

18 Apr '07

Hi all, I have problems with chip Realtek ALC883 (lspci: 00:06.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)) and surround sound. Stereo sound is working (games, music) fine - all 6 speaker are playing. But when I tried to run "aplay chan-id.wav", i've heard only "front left" (front & rear left + center were playing) and "front right" (right front & rear were playing), then was silence until the end of the file. Running "aplay -Dsurround51 chan-id.wav" causes that aplay hangs-up without any output. Running it with -v parameter produces this output: Playing WAVE 'Desktop/Surround-SDL-testfiles/chan-id.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Channels 6 Soft volume PCM Control: PCM Playback Volume min_dB: -51 resolution: 256 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 Slave: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 and then hang up until i press ctrl+c. Same results i get with speaker-test. I'm using alsa-lib-1.0.14-0.4.rc3.fc7 I've found similar problem in archive (about half a year old) but it still seems it isn't working. Is there any possibility, that this will be fixed in future? Thanks for all your help (and please leave me in CC, i'm not on mailing-list). Following lsmod & /proc/asound/card0/codec#0 [cdome@andromeda ~]$ /sbin/lsmod |grep snd snd_hda_intel 27936 2 snd_hda_codec 262400 1 snd_hda_intel snd_seq_dummy 11908 0 snd_seq_oss 40448 0 snd_seq_midi_event 16128 1 snd_seq_oss snd_seq 60704 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 16404 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 50944 0 snd_mixer_oss 24320 2 snd_pcm_oss snd_pcm 90888 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 30088 2 snd_seq,snd_pcm snd 67624 12 snd_hda_intel,snd_hda_codec,snd_seq_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 17505 2 snd snd_page_alloc 16912 2 snd_hda_intel,snd_pcm [cdome@andromeda ~]$ cat /proc/asound/card0/codec#0 Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x1458e601 Revision Id: 0x100002 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x17 0x17] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x04] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01a19c40: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01813c41: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9933014f: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x01cb7160: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

3 8

[alsa-devel] snd_pcm_hardware_t.info question
by Ciaccia 18 Apr '07

18 Apr '07

Hi all, I am trying to fix (improve?) a driver for an embedded ARM device I bought months ago, and for some reasons some ALSA applications work fine while other ones don't... Looking at the driver I got, I noticed that the info field in the snd_pcm_hardware_t struct just defines: SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE while the info in the tutorial "writing an alsa driver" defines: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID Furthermore, the driver defines a copy callback (actually 2, one for playback and one for capture), where it copies the data from the use space to the dma buffer. Since the dma buffer was preallocated with snd_pcm_lib_preallocate_pages_for_all and is accessible from the "outside", I wonder why the original author did this... I have some questions for you gurus: - what is the difference between SNDRV_PCM_INFO_MMAP and SNDRV_PCM_INFO_MMAP_VALID? What is SNDRV_PCM_INFO_BLOCK_TRANSFER? - The code calls snd_pcm_lib_preallocate_pages_for_all function as following: /* allocate the pcm(DMA) memory */ ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,0, 4*128*1024, 4*128*1024); is the DMA_TYPE right for an ARM device? Is NULL a correct value for data in this case? - In case I get rid of the copy callback, how do I specify the format of the stream? How does Where can I find an example for that? Thanks a lot Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 5

[alsa-devel] [PATCH] sound: convert "sound" subdirectory to UTF-8
by John Anthony Kazos Jr. 17 Apr '07

17 Apr '07

From: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> Convert the "sound" subdirectory to UTF-8. The following files are changed. sound/oss/{trident.c,es1371.c,pas2_pcm.c} sound/oss/emu10k1/{passthrough.h,passthrough.c} sound/pci/mixart/mixart.c Signed-off-by: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> --- I can't get my mail client to send in ISO-8859-1 explicitly instead of whatever it decides I want, so the actual patch is attached in octet-stream format. The below patch is just for reference and will very likely NOT work, unless this time it picks ISO-8859-1. Also, since the patch includes both encodings, whichever encoding you select while viewing it will display one correct and one garbled line. However, the bytecodes -are- correct. diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.c -- Emu10k1 digital passthrough - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.h -- Emu10k1 digital passthrough header file - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/es1371.c linux-2.6.21-rc7.mod/sound/oss/es1371.c --- linux-2.6.21-rc7.orig/sound/oss/es1371.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/es1371.c 2007-04-17 13:55:45.000000000 -0400 @@ -100,7 +100,7 @@ * Tjeerd Mulder <tjeerd.mulder(a)fujitsu-siemens.com> * 05.01.2001 0.29 Hopefully updates will not be required anymore when Creative bumps * the CT5880 revision. - * suggested by Stephan Müller <smueller(a)chronox.de> + * suggested by Stephan MÃ¼ller <smueller(a)chronox.de> * 31.01.2001 0.30 Register/Unregister gameport * Fix SETTRIGGER non OSS API conformity * 14.07.2001 0.31 Add list of laptops needing amplifier control diff -uprN linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c --- linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c 2007-04-17 13:55:45.000000000 -0400 @@ -85,7 +85,7 @@ static int pcm_set_speed(int arg) * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/. * * I cleared bit 5 of these values, since that bit controls the master - * mute flag. (Olav Wölfelschneider) + * mute flag. (Olav WÃ¶lfelschneider) * */ #if !defined NO_AUTO_FILTER_SET diff -uprN linux-2.6.21-rc7.orig/sound/oss/trident.c linux-2.6.21-rc7.mod/sound/oss/trident.c --- linux-2.6.21-rc7.orig/sound/oss/trident.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/trident.c 2007-04-17 13:55:45.000000000 -0400 @@ -18,7 +18,7 @@ * Ollie Lho <ollie(a)sis.com.tw> SiS 7018 Audio Core Support * Ching-Ling Lee <cling-li(a)ali.com.tw> ALi 5451 Audio Core Support * Matt Wu <mattwu(a)acersoftech.com.cn> ALi 5451 Audio Core Support - * Peter Wächtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support + * Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support * Muli Ben-Yehuda <mulix(a)mulix.org> * * @@ -89,7 +89,7 @@ * use set_current_state, properly release resources on failure in * trident_probe, get rid of check_region * v0.14.9c - * August 10 2001 Peter Wächtler <pwaechtler(a)loewe-komp.de> + * August 10 2001 Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> * added support for Tvia (formerly Integraphics/IGST) CyberPro5050 * this chip is often found in settop boxes (combined video+audio) * v0.14.9b diff -uprN linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c --- linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c 2007-04-17 13:55:45.000000000 -0400 @@ -472,7 +472,7 @@ static int snd_mixart_prepare(struct snd struct snd_mixart *chip = snd_pcm_substream_chip(subs); struct mixart_stream *stream = subs->runtime->private_data; - /* TODO de façon non bloquante, réappliquer les hw_params (rate, bits, codec) */ + /* TODO de faÃ§on non bloquante, rÃ©appliquer les hw_params (rate, bits, codec) */ snd_printdd("snd_mixart_prepare\n");

1 0

[alsa-devel] Request for CS4280 Manual
by Amar singh 17 Apr '07

17 Apr '07

Hello Sir, I'm a newbie in Linux Deivce Drivers and Kernel Development. We're writing our own driver for sound card - Cirrus Logic, Sound Fusion CS4280. in Fedora Core 6 Linux flavour. For that I've been looking for the programming manual of the above said sound card, for a long time now, but didn't succeed in getting one. Since nothing being fruitful for me in this case, we're stuck up with our project, at the start. Therefore I'd kindly like to request you to plz, send me, if you can, the programming manual in the pdf/doc format or a link to the place from where we can get that document. We'd really appreciate your help for the matter. Looking ahead for positive reply from you. Thanks and Regards, Amar singh Emertxe Information Technologies (P) Ltd., Bangalore - 56. India.

3 3

[alsa-devel] [PATCH] ASoC Kconfig description.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch makes the ASoC Kconfig descriptions a little more meaningful. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 1

[alsa-devel] Why do I not see patches in the emails?
by Timur Tabi 17 Apr '07

17 Apr '07

Can someone tell me why I can't see that actual patches in the posts that include them? I'm guessing it's some odd configuration option, but I went to http://mailman.alsa-project.org/mailman/options/alsa-devel and I didn't see anything applicable. Here's an example. As you can see, there is a text/plain attachment where the patch would be, but it's empty. Sender: alsa-devel-bounces(a)alsa-project.org Errors-To: alsa-devel-bounces(a)alsa-project.org Return-Path: alsa-devel-bounces(a)alsa-project.org X-OriginalArrivalTime: 16 Apr 2007 13:52:51.0309 (UTC) FILETIME=[864861D0:01C7802E] --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain Content-Transfer-Encoding: 7bit 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> --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel --=-Uhr6lvaXHMfZKXRJXsBj-- -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] FW: Individual PCM stream control
by Nagalla, Hari 17 Apr '07

17 Apr '07

Thanks for the reply.. We followed your suggestions and we see the PCM stream controls from the user space after we have created the controls at the driver using IFACE_PCM tags.. But when I try to set the 'index' in 'snd_ctl_elem_id' to reflect the PCM substream number, it seems to always get reset to '0'. Can you please suggest, how to set the index. --- snapshot of the code -------- snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_numid(id, ctl_num); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_PCM); fprintf(stdout,"setting the mute/unmute for stream index 0x%x \n", (unsigned int) stream_id); snd_ctl_elem_id_set_index(id,(unsigned int) stream_id); if (handle == NULL && (err = snd_ctl_open(&handle, card, 0)) < 0) { fprintf(stderr,"Control %s open error: %s\n", card, snd_strerror(err)); return err; } snd_ctl_elem_info_set_id(info, id); if ((err = snd_ctl_elem_info(handle, info)) < 0) { fprintf(stderr,"Cannot find the given element from control %s\n", card); return err; } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(control, id); show_control_id(id); printf("\n"); fprintf(stdout, "Retrieving the stream index 0x%x \n",snd_ctl_elem_id_get_index(id)); ---------------------------------------------------------------- Thanks a bunch !! Hari > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, April 04, 2007 11:21 AM > To: Nagalla, Hari > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Individual PCM stream control > > At Fri, 30 Mar 2007 05:47:11 -0500, > Nagalla, Hari wrote: > > > > Thanks for the advice.. Couple of inline questions below.. > > > > > > > -----Original Message----- > > > > I am new to ALSA, so some help from experienced folks would help.. > > > > > > > > We are trying to develop ALSA support for our DSP based mixer. In > > our > > > > architecture DSP performs the mixing, sample rate conversion etc. My > > > > question is how, can does the user space applications exercise > > > > controlling individual PCM stream to the mixer? Ex: individual > > stream > > > > gain control, individual stream mute, etc.. Any user space example > > code > > > > would be useful too.. > > > > > > Most drivers create PCM volume controls with IFACE_PCM tag, and set up > > > the volume appropriately at each PCM open, or let alsa-lib configs to > > > set up the initial volumes. Usually, the control element index > > > corresponds to the PCM substream number. > > > > Which alsa library API lets us to set up the PCM stream volume? And can > > it be used dynamically when the stream is active? > > It is controlled via the normal control API. Open the snd_ctl_t > handle for the same card (e.g. "hw:0" or "default"), and change the > value of the element. However, there is no standard name for the > individual PCM volume controls, and currrently depending on the driver > implementation... > > > I could n't figure out 'index' of the control element. > > It's the index in snd_ctl_elem_id. You can access via > snd_ctl_elem_id_{get|set}_index() in alsa-lib. > > > Takashi

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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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[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>

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[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>

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[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>

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[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>

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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

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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

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[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

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[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…

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[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

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[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.

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[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 }

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[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 }

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[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

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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

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[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!

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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-----

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[alsa-devel] Functioning support for Prodigy 192
by dustin＠seznam.cz 12 Apr '07

12 Apr '07

Hi Takashi, I am finishing changes to prodigy192.c to make the card functional. I modified eeprom values (e.g. 49MHz crystal), fixed digital output of the card (requires GPIO), added support for MI/ODI/O and added input switch mic/line. So far I have not done any MIDI tests. Before posting corresponding patches I would like to resolve a few remaining issues. The ak4114 based miodio card is a separately sold add-on and provides SPDIF out (directly connected to SPDIFOUT pin of ice1724), SPDIF IN (TOSLINK to input 0, coax to input 1 of ak4114) + MIDI IN/OUT. SPDIF OUT required no code, SPDIF IN is fully working with my last patches, switching between toslink/coax works in my prodigy192.c. Windows drivers detect miodio automatically, offering corresponding controls when connected. What would be a proper way in the driver to activate the miodio support? There is one Xilinx FPGA chip onboard, perhaps it is used for MIDI (although ice1724 has a built-in support, right?), perhaps it informs original windows drivers about version and capabilities of the addon card. Another option would be to test communication with ak4114, whether we can read contents of a particular register. Perhaps a proper place to do so would be in snd_ak4114_create of ak4114.c which could return error when no communication could be established with ak4114 (I do not think this is the case now). What solution would you suggest? Another question concerns capture input switches, both analog and digital. For now I have used a single switch SNDRV_CTL_ELEM_TYPE_BOOLEAN for each. On the capture tab of alsamixer that is not intuitive (capture off = mice, capture on = line etc.). I would prefer standard "capture input radio buttons" like e.g. in ac97 codec (ich5 + AD1985). I would need two independent groups of two "radio buttons". Please could you point me to some self-explaining code implementing similar feature in another driver which a newbie could understand? Thanks a lot. Pavel.

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[alsa-devel] ALSA Wiki
by Jaroslav Kysela 12 Apr '07

12 Apr '07

Hello all, new Wiki running well known MediaWiki (wikipedia) which will replace ALSA WWW contents is prepared at URL http://bugtrack.alsa-project.org/main/ . I would like ask possible volunteers to start moving the current contents of ALSA WWW server to this Wiki. Also, if you like to be a WikiSysop to change at least some Wiki layout - you have a chance now. Also, we need sysops to rollback wrong changes. The edit policy is simple. You have to be logged in and you MUST acknowledge your e-mail address. I'm sorry, but we cannot take contents from anonymous users. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Unblock MIDI IN in blocking read mode
by Florian 12 Apr '07

12 Apr '07

Hi, for best performance, I use a blocking snd_rawmidi_read() on an ALSA MIDI input device, byte by byte. It works fine, except: 1) How do I close the device from a different thread? I.e. how can I unblock the blocking read() call when it's time to close the device and no MIDI data is currently coming in? Just calling snd_rawmidi_close() will cause a crash. Using snd_rawmidi_drop() did not release the blocking read() in the other thread. 2) How is a blocking read() typically implemented in the ALSA MIDI driver? Maybe I can resort to non-blocking mode and avoid the situation above. 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!

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[alsa-devel] [PATCH] ALSA: Support Media Vision Jazz16 chips
by Rask Ingemann Lambertsen 11 Apr '07

11 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add support for Jazz16 features to the common SoundBlaster code. - Add a front end for Jazz16 based sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch is a prerequisite for the Jazz PnP patch, which needs the patched include/sound/sb.h. Documentation/sound/alsa/ALSA-Configuration.txt | 10 include/sound/sb.h | 5 sound/isa/Kconfig | 15 + sound/isa/sb/Makefile | 2 sound/isa/sb/jazz16.c | 193 ++++++++++++++ sound/isa/sb/sb8_main.c | 105 +++++++ sound/isa/sb/sb_common.c | 12 sound/isa/sb/sb_mixer.c | 3 8 files changed, 335 insertions(+), 10 deletions(-) --- linux-2.6.20/sound/isa/sb/Makefile.orig 2007-02-19 21:00:09.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/Makefile 2007-03-02 20:14:48.000000000 +0100 @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o --- linux-2.6.20/sound/isa/sb/jazz16.c-orig 2007-02-17 20:01:40.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/jazz16.c 2007-03-18 13:43:35.000000000 +0100 @@ -0,0 +1,193 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * There is also an MPU-401 interface. + * + * The IBM PPS Model 6015 has a Jazz16 chip on board. Please tell me if it + * works with this driver or not. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +MODULE_AUTHOR ("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION ("Jazz16"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Jazz16 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Jazz16 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Jazz16 soundcard."); + +#define PFX "jazz16: " +#define PFX_WARN KERN_WARNING PFX +#define PFX_ERR KERN_ERR PFX + +static struct pnp_driver snd_jazz16_pnp_driver; + +static irqreturn_t jazz16_interrupt(int irq, void *dev_id) +{ + struct snd_sb *chip = (struct snd_sb *) dev_id; + return snd_sb8dsp_interrupt(chip); +} + +static struct pnp_device_id snd_jazz16_pnpids[] = { + { .id = "PNPb00f" }, + { .id = "" } +}; +MODULE_DEVICE_TABLE(pnp, snd_jazz16_pnpids); + +static int __devinit snd_jazz16_pnp_probe(struct pnp_dev *pnp_dev, + const struct pnp_device_id *pnp_id) +{ struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int err; + uint sbport, sbirq, sbdma8, sbdma16, mpuport, mpuirq; + static uint dev_num = 0; + + if (enable[dev_num]) + card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); + else + card = NULL; + dev_num ++; + if (card == NULL) + return -ENOMEM; + pnp_set_drvdata(pnp_dev, card); + /* TODO use pnp_port_valid(), pnp_port_flags(), pnp_port_length()... */ + sbport = pnp_port_start(pnp_dev, 0); + sbirq = pnp_irq(pnp_dev, 0); + sbdma8 = pnp_dma(pnp_dev, 0); + sbdma16 = pnp_dma(pnp_dev, 1); + err = snd_sbdsp_create(card, sbport, sbirq, jazz16_interrupt, + sbdma8, sbdma16, SB_HW_AUTO, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + /* TODO length limits - strncpy()/snprintf() and friends. */ + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at 0x%x, irq %u, dma8 %u, dma16 %u", + chip->name, sbport, sbirq, sbdma8, sbdma16); + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(PFX_ERR "Not a Jazz16 chip at 0x%x.\n", sbport); + snd_card_free(card); + return -ENODEV; + } + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sbmixer_new(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_opl3_create(card, sbport, sbport + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) { + snd_printk(PFX_WARN "No OPL device found, skipping.\n"); + } else { + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + } + if (pnp_port_valid(pnp_dev, 1) + && !(pnp_port_flags(pnp_dev, 1) & IORESOURCE_DISABLED)) { + int mpuirqflags; + if (!pnp_irq_valid(pnp_dev, 1) + || pnp_irq_flags(pnp_dev, 1) & IORESOURCE_DISABLED) { + mpuirq = -1; + mpuirqflags = 0; + } else { + mpuirq = pnp_irq(pnp_dev, 1); + mpuirqflags = IRQF_DISABLED; + } + mpuport = pnp_port_start(pnp_dev, 1); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport, 0, mpuirq, + mpuirqflags, NULL); + if (err < 0) { + snd_printk(PFX_WARN "MPU-401 device not configured, skipping.\n"); + } + } + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + return 0; +} + +static void __devexit snd_jazz16_pnp_remove(struct pnp_dev *dev) +{ + snd_card_free(pnp_get_drvdata(dev)); + pnp_set_drvdata(dev, NULL); +} + +static struct pnp_driver snd_jazz16_pnp_driver = { + .name = "Jazz16", + .id_table = snd_jazz16_pnpids, + .probe = snd_jazz16_pnp_probe, + .remove = __devexit_p(snd_jazz16_pnp_remove), +}; + +static int __devinit alsa_card_jazz16_init (void) +{ + return pnp_register_driver(&snd_jazz16_pnp_driver); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + pnp_unregister_driver(&snd_jazz16_pnp_driver); +} + +module_init (alsa_card_jazz16_init); +module_exit (alsa_card_jazz16_exit); --- linux-2.6.20/sound/isa/sb/sb_mixer.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb_mixer.c 2007-02-25 01:09:05.000000000 +0100 @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: --- linux-2.6.20/sound/isa/sb/sb_common.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb_common.c 2007-03-03 12:50:32.000000000 +0100 @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; --- linux-2.6.20/sound/isa/sb/sb8_main.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb8_main.c 2007-03-24 23:30:22.000000000 +0100 @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Sat Mar 24 23:29:20 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements (ported from sound/oss/sb_common.c). */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,14 +113,53 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + if (c->max > 1) { + unsigned int num = 0, den = 0; + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -117,6 +168,7 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; @@ -134,8 +186,17 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + } else { + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -173,7 +234,7 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -232,9 +293,11 @@ static int snd_sb8_capture_prepare(struc unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -242,6 +305,8 @@ static int snd_sb8_capture_prepare(struc chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } + /* fall through */ + case SB_HW_JAZZ16: chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO; break; case SB_HW_201: @@ -261,10 +326,17 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + } else { + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -285,7 +357,7 @@ static int snd_sb8_capture_prepare(struc snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -358,10 +430,12 @@ static snd_pcm_uframes_t snd_sb8_playbac { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_PLAYBACK_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -369,10 +443,12 @@ static snd_pcm_uframes_t snd_sb8_capture { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_CAPTURE_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -443,6 +519,19 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; --- linux-2.6.20/sound/isa/Kconfig.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/Kconfig 2007-03-25 21:49:44.000000000 +0200 @@ -248,6 +248,21 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Media Vision Jazz16 based + soundcards. + + You probably want to enable Jazz16 Plug and Play support + (CONFIG_JAZZ16PNP) also. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND --- linux-2.6.20/Documentation/sound/alsa/ALSA-Configuration.txt-orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/Documentation/sound/alsa/ALSA-Configuration.txt 2007-03-25 20:35:31.000000000 +0200 @@ -1120,6 +1120,16 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based on Media Vision Jazz16 chip (PnP only) + + This module supports multiple cards and PnP. + + Note: You will most likely want to enable Jazz16 PnP support + (CONFIG_JAZZ16PNP) in the kernel. + Module snd-korg1212 ------------------- --- linux-2.6.20/include/sound/sb.h.orig 2007-02-19 20:57:51.000000000 +0100 +++ linux-2.6.20/include/sound/sb.h 2007-03-25 21:37:01.000000000 +0200 @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ @@ -142,6 +143,7 @@ struct snd_sb { #define SB_DSP_HI_INPUT_AUTO 0x98 #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +267,9 @@ struct snd_sb { #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ -- Rask Ingemann Lambertsen

3 15

[alsa-devel] snd_mixer_selem_channel_id -- should i be using this? if not, what should i use?
by John L. Utz III 11 Apr '07

11 Apr '07

Hello Everybody; mixer.h indicates that snd_mixer_selem_channel_id is a legacy api. so, what is the non legacy api for setting volumes on individual channels, ie Master L&R, PCM L&R and Surround L&R ? tnx! johnu

1 0

[alsa-devel] Interrupts and kmalloc in alsa modules
by Ciaccia 11 Apr '07

11 Apr '07

Hi all, I am working with an ARM embedded system and I porting an alsa application to that system. Cirrus (the manufacturer of the board) wrote a generic alsa driver for "their" kernel (a kernel based on the 2.6.8 with some patches) and I ported the module to the vanilla kernel 2.6.19. The module works fine (play and capture both work), but the function that allocates the memory prints a strange message in the kernel log when the buffers are allocated (the module calls kmalloc, which prints "sleeping function called from invalid context" to the log). The buffers are allocated when the streams are started, not when the module is loaded, I think this is standard... The buffers are allocated with the following line: s->buffers = (audio_buf_t *) kmalloc(sizeof(audio_buf_t) * s->nbfrags, GFP_KERNEL); When the line here above is executed, slab.c prints the error the kernel log. The error comes from the "might_sleep_if" line: static inline void cache_alloc_debugcheck_before(struct kmem_cache *cachep, gfp_t flags) { might_sleep_if(flags & __GFP_WAIT); #if DEBUG kmem_flagcheck(cachep, flags); #endif } Playing with the module, I discovered that if I enable the interrupts before calling kmalloc, the error disappears. I wonder if this is correct, or if something is wrong... Could someone of you explain to me why I have to enable the interrupts before calling kmalloc? The strange thing is that the module never explicitly turns them off (I attached the file to the mail) Any help is greatly appreciated ;-) Bye Andrea ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

3 5

[alsa-devel] sound on asus p5ld2-se
by Andrius 10 Apr '07

10 Apr '07

Hi, It says "If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel ML" in ALSA-configuration.txt so, here it is: i'm using asus p5ld2-se mobo with intel ich7 chipset, and just have installed new debian stable (4.0). And encountered a problem: onboard audio sound from one of the speakers was just static noise, and if i adjusted volume or anything in alsamixer, the sound kept dissappearing. I tried module=3stack option in modprobe, added line options snd-hda-intel index=0 model=3stack in /etc/modprobe.d/sound and everything is ok now. i attached lspci -nv output to this letter p.s. on ubuntu live cd and other distros i've tried there were same problem

2 1

[alsa-devel] Driver for user-space sound source/sink(s)
by Alex Makarov 10 Apr '07

10 Apr '07

Hi We're considering implementing the driver for user-space sound input/output. For example: "Line in" is provided by some application by streaming data to a file in /dev/..., "Line out" can be read by any application from /dev/... It would be also nice to have arbitrary number of streams, for example to have different applications to "capture" audio from different sources. I wonder if we reinvent the wheel, or maybe there's an ongoing project for the similar driver. If not, could anyone advise how this can be achieved. We had no experience with ALSA before, but pretty strong in C/C++ development. Thanks! Alex

1 0

[alsa-devel] Volume control channels mixed
by Simon van der Linden 10 Apr '07

10 Apr '07

Hello, I use Alsa in the latest Ubuntu bêta, with my MacBook. lspci gives the audio card as: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Sound is working pretty good. However, volume control channels (in alsamixer) are mixed: with laptop speakers: Master controls treble volume Front controls bass volume with headphones: Master has no effect Front controls the volume I'm unfamiliar with Alsa, but I guess it shouldn't be too hard to correct that. Browsing the web didn't give me more informations. Perhaps some of you could help me? -- Simon van der Linden jid: simon(a)vanderlinden.eu.org

2 3

Re: [alsa-devel] [Audacity-devel] distortion with alsa I/O - no distortion with oss emulation
by Simon Lewis 10 Apr '07

10 Apr '07

Dear Alsa developers Audacity seems to work perfectly with AOSS and Jack but not directly with ALSA on some linux PCs. Trying to use ALSA directly leads to severe distortion due to lack of sync with Alsa. (I hope that is the correct terminology.) I received this reply from the Audacity developers, what tools are available from ALSA to investigate this problem? How can I determine what the real "Alsa configuration" is that portaudio has used setting up communication with Alsa. Best Regards, Simon > Markus Meyer schrieb: > > > As for Simon's problems with "synchronisation", I'm not so sure that it > is a Portaudio issue, or a software issue at all. Software issues tend > to be of the "on/off" type, and Portaudio does work well with ALSA in > the general case. Apart from being a hardware issue (defective sound > card, main board or something) this could also well be a race condition > in the ALSA driver (kernel module) which is only triggered by certain > buffer sizes, order of driver requests etc. Because Audacity is > multi-threaded it certainly stresses the driver/kernel subsystem more > than just, say, "arecord", which just gets blocks from the driver and > writes it out to disk. > > > Markus > > >> Simon Lewis schrieb: >> >> The main issue is that on some linux machines, mine included, there is >> severe distortion when Audacity is used with ALSA drivers. >> >> You will be pleased to hear that I have found the root cause of the >> problem. It is nothing to do with the options with which Audacity is >> compiled. The root cause is that portaudio loses synchronisation with >> ALSA's data stream. >> >> I was able to verify this yesterday, as I could start Audacity with >> Jack, and through Jack set up a number of simulation conditions and >> monitor the results. >> >> With my standard Jack settings of 96kHz / 256 Frames per period / 2 >> periods per buffer (latency 5.33 msecs.), ALSA initialises with sample >> format 32bit little-endian, 2 periods for capture and 2 periods for >> playback. >> >> Under these conditions Audacity cannot synchronise with the data stream >> and severe distortion is the result. >> >> However, increasing the frames per period to 1024 (latency 21.3 msec) >> and Audacity (1.3.2 with resampling disabled) runs perfectly and I can >> record for a long time. (CPU load is 9.6%) >> >> Reducing the frames per period to 512 and the same version of Audacity >> starts OK but after a few seconds, sometimes longer losses sync with the >> data stream. >> >> The standard 1.3.2 version of Audacity with resampling enabled does not >> sync on start with 512 frames per period but runs OK with Jack set to >> 1024 frames per period. >> >> The 1.3.3 version of Audacity complied with libsamplerate without >> portmixer runs OK with 512 frames per period under jack. >> >> NONE of the Audacity versions will run directly with the ALSA drivers. >> The Version with resampling disabled often starts OK, but after 30 sec >> to a few minutes of recording sync is lost and naturally distortion is >> the result. >> >> The conclusion is that portaudio is very poor at syncing with the >> alsadata stream, something which Jack can do with very little buffering. >> >> >> Ciao, Simon

3 2

[alsa-devel] Correct use of ak4114.c?
by dustin＠seznam.cz 10 Apr '07

10 Apr '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O card, I used corresponding code from revo.c (and similarly from juli.c), calling function snd_ak4114_create. However, this function does not initialize array ak4114->kctls[idx], leading to consequent kernel oops after the periodically called snd_ak4114_check_rate_and_errors, specifically in function snd_ctl_notify called by snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4114->kctls[0]->id) Temporarily I resolved the problem by commenting out the snd_ctl_notify calls. However, it seems author of ak4114.c intended to create the device with snd_ak4114_build instead which defines all the controls. I did not hit the problem until fixing the 4-wire communication in revo.c - that explaing why nobody had this problem with revo cards. I do not understand how the code can work in juli.c - perhaps I have overlooked something there. My question: Is the correct way to add null pointer checks before snd_ctl_notify calls only, or to use snd_ak4114_build insted of snd_ak4114_create? If using snd_ak4114_build, where do I get the parameters ply_substream and cap_substream? int snd_ak4114_build(struct ak4114 *ak4114, struct snd_pcm_substream *playback_substream, struct snd_pcm_substream *capture_substream); Thanks for any suggestions. Pavel.

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[alsa-devel] seq: timing unstable on slow computer when busy
by Gerald Grabner 09 Apr '07

09 Apr '07

Hi there, I'm trying to use a simple (self written) sequencer application on an old laptop with a 350 MHz CPU. It works, but the timing of output events becomes unstable when my program is busy with things like scheduling events for the subsequent loop or an unmuted track. I'm wondering about the best approach to tackle this problem. Depending on how the priority of the kernel's event scheduling is related to the priority of an application, I could imagine some of the following measures: * Running my process with lower or higher priority * Using (priorized) threads for the event scheduling * Setting Linux kernel config options * Setting ALSA related options (driver or lib) * ... While playing midi, I'm scheduling an echo event a few ticks before the end of every sequence, upon which a function to schedule the events of the next sequence is called. What is the typical length of this scheduling span: Just one bar? Or longer? Many thanks in advance, Gerald BTW: My operation system is Gentoo's Linux 2.6.19 with alsa-driver 1.0.14_rc1.

2 1

[alsa-devel] Popping noise on HP dv6000z (CX20549) running x86 kernel
by enigma_0Z 08 Apr '07

08 Apr '07

Here's my situation: If I play a game (any game, in particular, however, neverball/putt or sauerbraten), the sound has these little soft "pop" noises happening so quick per second I cannot count them. This is when I'm running on an x86 kernel, however, when I run an x86_64 kernel, these are not present. I've got a 64-bit processor, specifically an AMD Sempron 3500+ Does Anyone know why? I've got a conexant CX20549 in my laptop. Thanks, enigma

1 1

[alsa-devel] dont knw what this error means
by Ashlesha Shintre 05 Apr '07

05 Apr '07

Hi, I am getting the following error when I try to playback a raw sample file: "ALSA lib pcm.c:7216snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Does anyone know what this means? Thanks Ashlesha.

2 1

[alsa-devel] alsa+saa7134 sound problem
by aa 05 Apr '07

05 Apr '07

please read marco007 posts my english is so bad http://forums.gentoo.org/viewtopic-t-527329-postdays-0-postorder-asc-start-… tnx

2 1

[alsa-devel] Sampling rate limited to 48 kHz
by Marek Ławicki 05 Apr '07

05 Apr '07

Hi, I have Turtle Beach Catalina Card (Envy24HT-S, VT1616 + WM8728) which I belive is exactly the same as Mad Dog Entertainer 7.1 ( http://techreport.com/reviews/2004q1/maddog-entertainer/index.x?pg=2 ) Card is recognized as Chaintech AV-710 but works very well ( so far I tested analog out through WM8728 and digital out ). My problem is : I can't set sampling rate higher than 48000 on SPDIF out. I know this card can go higher because : 1. It can go higher under Windows (High Sample Rate 2-channel mode) 2. If I "flash" eeprom with M-Audio Revolution 7.1 settings card can sync with my external DAC up to 96 kHz (DAC limit) and I can play 96kHz .wav file. This is not the way to go for me (analog part and SPDIF-in is lost) I found this : ice1724.c:390:static int get_max_rate(struct snd_ice1712 *ice) ice1724.c-391-{ ice1724.c-392- if (ice->eeprom.data[ICE_EEP2_ACLINK] & VT1724_CFG_PRO_I2S) { ice1724.c-393- if ((ice->eeprom.data[ICE_EEP2_I2S] & 0x08) && !ice->vt1720) ice1724.c-394- return 192000; ice1724.c-395- else ice1724.c-396- return 96000; ice1724.c-397- } else ice1724.c-398- return 48000; ice1724.c-399-} ice1724.c-400- Tried to play with it, but no luck - I'm not developer :-) How can I help with making ice1724 driver even better ? Thanks, Marek # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc1. Compiled on Apr 2 2007 for kernel 2.6.18.6. # cat /proc/asound/card0/ice1724 Chaintech AV-710 at 0xd000, irq 11 EEPROM: Subvendor : 0x12142417 Size : 28 bytes Version : 1 System Config : 0x42 ACLink : 0x2 I2S : 0x10 S/PDIF : 0xc3 GPIO direction : 0xff GPIO mask : 0xff GPIO state : 0xff Extra #18 : 0x1 Extra #19 : 0x8 Extra #20 : 0x0 Extra #21 : 0x0 Extra #22 : 0x0 Extra #23 : 0x0 Extra #24 : 0x0 Extra #25 : 0x0 Extra #26 : 0x0 Extra #27 : 0x0 Registers: PSDOUT03 : 0x00000000 CCS00 : 0x00 CCS01 : 0xa0 CCS02 : 0x00 CCS03 : 0x00 CCS04 : 0x42 CCS05 : 0x02 CCS06 : 0x10 CCS07 : 0xc3 CCS08 : 0x00 CCS09 : 0x00 CCS0a : 0x00 CCS0b : 0x00 CCS0c : 0x00 CCS0d : 0x0a CCS0e : 0x00 CCS0f : 0x00 CCS10 : 0x37 CCS11 : 0x06 CCS12 : 0x01 CCS13 : 0x80 CCS14 : 0x00 CCS15 : 0xea CCS16 : 0xff CCS17 : 0x00 CCS18 : 0xff CCS19 : 0x00 CCS1a : 0x00 CCS1b : 0x00 CCS1c : 0x00 CCS1d : 0x00 CCS1e : 0x2a CCS1f : 0x00 MT00 : 0x00 MT01 : 0x08 MT02 : 0x00 MT03 : 0x08 MT04 : 0x5a MT05 : 0x08 MT06 : 0x21 MT07 : 0x82 MT08 : 0x00 MT09 : 0x00 MT0a : 0x00 MT0b : 0x00 MT0c : 0x00 MT0d : 0x00 MT0e : 0x00 MT0f : 0x00 MT10 : 0x40 MT11 : 0x1f MT12 : 0xa8 MT13 : 0x0d MT14 : 0x2f MT15 : 0x78 MT16 : 0x00 MT17 : 0x00 MT18 : 0x80 MT19 : 0x03 MT1a : 0x00 MT1b : 0x00 MT1c : 0xff MT1d : 0x07 MT1e : 0x00 MT1f : 0x00 MT20 : 0x20 MT21 : 0x01 MT22 : 0xac MT23 : 0x0d MT24 : 0xb7 MT25 : 0x7f MT26 : 0xff MT27 : 0x07 MT28 : 0x00 MT29 : 0x00 MT2a : 0x00 MT2b : 0x00 MT2c : 0x00 MT2d : 0x00 MT2e : 0x00 MT2f : 0x00 # lspci -vvv 00:0b.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: VIA Technologies Inc. AMP Ltd AUDIO2000 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64 Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at d000 [size=32] Region 1: I/O ports at cc00 [size=128] Capabilities: [80] Power Management version 1 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME-

2 5

Re: [alsa-devel] [Alsa-devel] Quality resampling code for libasound
by Takashi Iwai 05 Apr '07

05 Apr '07

At Wed, 21 Mar 2007 08:35:22 +1100, Jean-Marc Valin wrote: > > >> Well, I use that type because that's what is used in the API. And the > >> reason I use it in the API is because this code is meant to work even on > >> platforms with 16-bit integers (e.g. TI C5X DSPs) and no C99 compiler. > >> If that's causing problem, I guess you could still use unsigned int with > >> no problem... > > > > Don't worry, alsa-lib would be very unlikely ported to such an > > architecture. > > Well, my code has to run on those architectures, and it's much simpler > if I don't need to change it for ALSA. Now if you don't want the > spx_uint32_t in the plugin code, then feel free to change that. OK, as long as it's in alsa-plugins repository, I basically don't care. But, for merging it to alsa-lib, I'd like to change the code very much (including coding-style unification). Anyway, I applied your latest codes to HG tree now (with the removal of samplerate.h). Thanks! Takashi

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Johannes Berg wrote: > On Wed, 2007-04-04 at 20:42 +0200, Stefan de Konink wrote: > >>> /* >>> * For input, we need to do a graceful stop; if we abort >>> * the DMA, we end up with leftover bytes that corrupt >>> * the next recording. To do this we set the S0 status >>> * bit and wait for the DMA controller to stop. Each >>> * command has a branch condition to >>> * make it branch to a stop command if S0 is set. >>> * On input we also need to wait for the S7 bit to be >>> * set before turning off the DMA controller. >>> * In fact we do the graceful stop for output as well. >>> */ >> Check, I have that. > > Hm. Then I don't know what could be causing this. Does it happen all the > time or just intermittently? With every OSS app out there... but native alsa works fine. Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGFBwQYH1+F2Rqwn0RCi2dAJ9Vlel3cz0PAUMvgYi7L2Nd3XCc9wCeJG9H NT+AsNRL91Dm1y9Dlch75PU= =xxif -----END PGP SIGNATURE-----

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Johannes Berg wrote: > On Wed, 2007-04-04 at 20:31 +0200, Stefan de Konink wrote: > >> In lsmod powermac is not loaded. The 'strange' thing I found, is that >> even without snd-pcm-oss or aoss I get garbage. At that moment I wonder >> what emulation layer is used. > > Oh, I see. There was a bug in the i2sbus setup, does the i2sbus-pcm.c > file (sound/aoa/soundbus/i2sbus/i2sbus-pcm.c) contain the following > comment at around line 380: Something I must note: recording works. Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGE/mcYH1+F2Rqwn0RCiYeAJ9h+F1tCdS/cyNbaXoVueFBXX09PgCdGfAp rRH4y2U/E/SBIHtwobJjVI4= =FdSb -----END PGP SIGNATURE-----

1 0

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 04 Apr '07

04 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Johannes Berg wrote: > On Wed, 2007-04-04 at 20:31 +0200, Stefan de Konink wrote: > >> In lsmod powermac is not loaded. The 'strange' thing I found, is that >> even without snd-pcm-oss or aoss I get garbage. At that moment I wonder >> what emulation layer is used. > > Oh, I see. There was a bug in the i2sbus setup, does the i2sbus-pcm.c > file (sound/aoa/soundbus/i2sbus/i2sbus-pcm.c) contain the following > comment at around line 380: > /* > * For input, we need to do a graceful stop; if we abort > * the DMA, we end up with leftover bytes that corrupt > * the next recording. To do this we set the S0 status > * bit and wait for the DMA controller to stop. Each > * command has a branch condition to > * make it branch to a stop command if S0 is set. > * On input we also need to wait for the S7 bit to be > * set before turning off the DMA controller. > * In fact we do the graceful stop for output as well. > */ Check, I have that. Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGE/F+YH1+F2Rqwn0RCtRLAJ0VTYJlaivU3gC0toB7cKI7AYBz6QCfV5AW BO+B/PhvRdxr+asrfI8f9uA= =0LCT -----END PGP SIGNATURE-----

1 0

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 04 Apr '07

04 Apr '07

At Wed, 04 Apr 2007 19:18:48 +0200, Johannes Berg wrote: > > On Wed, 2007-04-04 at 19:10 +0200, Takashi Iwai wrote: > > > > > To make sure, check /proc/asound/card0/pcm0p/sub0/hw_params has no > > > > "OSS" entry when you're playing via aoss. If a line like "OSS Format" > > > > appears there, it is still kernel oss-emulation. > > > > > > OSS Format is not in there. > > > > Fine. Then it smells more. > > > > Johannes, any known / possible problem with OSS emulation using aoa > > driver (especially regarding endianness)? > > snd-pcm-oss you mean? Both snd-pcm-oss and aoss wrapper seem not working for Stefan's machine. Maybe a problem specific to a certain model? Takashi > No, it's working fine, the contents of that file is (mplayer -ao oss): > > access: RW_INTERLEAVED > format: S16_BE > subformat: STD > channels: 2 > rate: 44100 (44100/1) > period_size: 1024 > buffer_size: 16384 > tick_time: 10000 > OSS format: S16_BE > OSS channels: 2 > OSS rate: 44100 > OSS period bytes: 4096 > OSS periods: 16 > OSS period frames: 1024 > > while with alsa it is (rhythmbox/gstreamer): > > access: MMAP_INTERLEAVED > format: S16_BE > subformat: STD > channels: 2 > rate: 44100 (44100/1) > period_size: 1024 > buffer_size: 8192 > tick_time: 10000 > > and with aoss (aoa mplayer -ao oss, snd-pcm-oss not loaded): > access: MMAP_INTERLEAVED > format: S16_BE > subformat: STD > channels: 2 > rate: 44100 (44100/1) > period_size: 1024 > buffer_size: 8192 > tick_time: 10000 > > > Don't see a problem at all, working well for me.

2 1

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 04 Apr '07

04 Apr '07

At Wed, 04 Apr 2007 19:12:49 +0200, Stefan de Konink wrote: > > Takashi Iwai wrote: > > I meant the comparison between the working state with ALSA apps and > > the non-working state with OSS apps... > > Lets put it this way: > mplayer -ao oss: noise > mplayer -ao alsa: nice sound And the comparison of parameters? You can get it in /proc/asound/card0/pcm0p/sub0/* files as I already mentioned. Diff them between two different stats. Takashi

1 0

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 04 Apr '07

04 Apr '07

At Wed, 04 Apr 2007 18:57:11 +0200, Stefan de Konink wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA512 > > Takashi Iwai wrote: > > Hm, then I suspect rather it's a problem in the aoa driver side. > > > > To make sure, check /proc/asound/card0/pcm0p/sub0/hw_params has no > > "OSS" entry when you're playing via aoss. If a line like "OSS Format" > > appears there, it is still kernel oss-emulation. > > OSS Format is not in there. Fine. Then it smells more. Johannes, any known / possible problem with OSS emulation using aoa driver (especially regarding endianness)? > > After you confirm it's a native ALSA in the driver execution, compare > > the parameters there between the working status (using ALSA API) and > > non-working status (OSS API). > > I never have had the aoa driver working with OSS. I'm using 1.0.14_rc3. I meant the comparison between the working state with ALSA apps and the non-working state with OSS apps... Takashi

1 0

[alsa-devel] Individual PCM stream control
by Nagalla, Hari 04 Apr '07

04 Apr '07

Hello, I am new to ALSA, so some help from experienced folks would help.. We are trying to develop ALSA support for our DSP based mixer. In our architecture DSP performs the mixing, sample rate conversion etc. My question is how, can does the user space applications exercise controlling individual PCM stream to the mixer? Ex: individual stream gain control, individual stream mute, etc.. Any user space example code would be useful too.. Thanks Hari

2 3

[alsa-devel] ALSA, GStreamer and surround
by Christian F.K. Schaller 04 Apr '07

04 Apr '07

Hi, We are still having issues resolving how to deal with Surround sound and ALSA in GStreamer. If anyone knowledgeable about ALSA would be willing to comment on this bug: http://bugzilla.gnome.org/show_bug.cgi?id=410403 We would much appreciate it. Christian

3 12

[alsa-devel] usb midi disconnect -> kernel oops
by Dmitry Baikov 04 Apr '07

04 Apr '07

Hi! If I have a running application which connects (via alsa seq) to usb midi device, and device is disconnected, I get a kernel oops. kernels tried: 2.6.20-rt8 (in-kernel alsa), 2.6.21-rc5-rt3 (in-kernel alsa, and 1.0.14_rc3) Kernel dump attached. Dmitry.

3 11

Re: [alsa-devel] SPDIF/IEC958 sample rate on HDA/ALC882
by Dag Lem 03 Apr '07

03 Apr '07

Takashi Iwai <tiwai <at> suse.de> writes: > > Hi, > > Sorry, looks like I overlooked this post... > > At Wed, 01 Nov 2006 17:58:55 +0100, > Erik Slagter wrote: > > > > Sorry if this is considered to be "user" question, but I got no response > > at all from the "user" list. Guess the real cracks hang around here. > > > > I have lots of questions I cannot resolve from all of the docs, but for > > the moment I'll stick to the most important one. > > > > The combination of HDA/ALC882 should be able to deliver sound at 96 kHz > > (hda..., right...) Also the ALC882 datasheet says it can deliver sound > > at 96 kHz. > > > > In practise, though, I cannot get it to output any other sample rate > > than 48 kHz. Any other sample rate handed to hw:0,0 is simply converted > > (somewhere??? and badly...) to 48 kHz and the SPDIF is still driven at > > 48 kHz. > > > > iecset allows me to set the rate to 32/44.1/48 kHz, the spdif output is > > still at 48 kHz. Some other flags from iecset are actually honoured > > (like "data" and "emphasis"). > > It's simply because iecset program doesn't support the rates over > 48kHz. But the hda-intel driver supports the rate, AFAIK. > (At least, there is no particular code that restricts over-48kHz.) > > One thing to be noted is that if you use "hw" PCM device, you have to > set up the SPDIF status bits _manually_ via control API. If you use > "iec958" or "spdif" PCM device, you can pass these bits as optional > arguments at opening the PCM. > > > I can ask aplay to play at various rates (from a suitable PCM file), but > > it complains at any other rate than 44.1 or 48 (notably 32) kHz that the > > rate is not supported. It plays at 44.1 kHz though. And again converts > > it to 48 kHz! Sigh... > > Who complains? At least, the driver won't. > > > Maybe I am using the wrong "hw", there is also a "hw:0,2" device, which > > I cannot make work properly at all (only one channel is output, large > > chunks are discarded, much much clipping). > > The first PCM device is for the multi-output PCM. It's for both > analog and digital. The dedicated SPDIF is the secondary one. > > Takashi > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > Hi, I was googling around and found this (old) post; I can confirm that there are severe problems with the ALC882 and/or the driver code. I'm experiencing the same problem as Erik - I cannot get the ALC882 to transmit at any other sample rate than 48kHz over S/PDIF. When I attempt to play 44.1kHz sound, the S/PDIF connection is still locked at 48kHz (shown by my amplifier), and the resulting sound is less than pleasing. It would seem like a 44.1kHz signal packed in a 48kHz frame rate played back directly - high-pitched sounds suffer badly from what my ears recognize as straightforward sample picking as opposed to resampling. Looking at hda_codec.c:convert_from_spdif_status() it seems that the IEC958 parameter bit for 44.1kHz/48kHz is not used for anything (correct me if I'm wrong). This would mean that the rate is only set in snd_hda_codec_setup_stream(), and as far as I can tell from looking at the array rate_bits and the ALC882 Series Datasheet the setting is correct for 44.1kHz. Since I can't find any obvious mistake in the HDA code, I guess there is some problem in patch_realtek.c, or an initialization incompatibility with the HDA code and the ALC882 chip. There are certainly initialization problems somewhere, as the S/PDIF output is sometimes not able to send AC3 data correctly (amplifier cannot detect AC3 and stays silent), and attempting to play audio (even 48kHz) results in intermittent screeches. This is solved by setting and resetting the IEC958 control widget, trying again and again until it finally works. When it does work, it works without a hitch until the next reboot. Any ideas? Is there anything I can help to test in order to nail down the problem? My motherboard is an Asus P5W DH Deluxe with an 82801G HDA controller, in case this has any relevance. -- Best regards, Dag Lem

4 20

[alsa-devel] Vanilla Blocked I/O probs
by Gordon Miller 03 Apr '07

03 Apr '07

Hi, I am calling rc = snd_pcm_readi (handle, buffer, MAX_LEN) but rc is always 4 short of MAX_LEN (1020 instead of 1024). I thought that it would just block until MAX_LEN frames were available. This is really doing my head in, and any help would be appreciated. Cheers

3 4

Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Takashi Iwai 03 Apr '07

03 Apr '07

At Tue, 03 Apr 2007 00:35:21 +0200, Stefan de Konink wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA512 > > Takashi Iwai wrote: > > At Wed, 28 Mar 2007 13:08:09 +0200 (CEST), > > Stefan de Konink wrote: > >> On Wed, 28 Mar 2007, Takashi Iwai wrote: > >> > >>> Did you try other drivers, such as usb-audio? > >> I did not. > > > > Then try it if you have one. > > Don't have one. It's a pity. > >>> And, you're using user-space alsa-oss wrapper (e.g. via aoss wrapper), > >>> right? The user-space emulation and kernel-space emulation are > >>> totally different, so I have to know what you actually tried. > >> I'm only using the in kernel translation. > > > > Then try aoss, too. > > aoss program gives the the same garbage. Hm, then I suspect rather it's a problem in the aoa driver side. To make sure, check /proc/asound/card0/pcm0p/sub0/hw_params has no "OSS" entry when you're playing via aoss. If a line like "OSS Format" appears there, it is still kernel oss-emulation. After you confirm it's a native ALSA in the driver execution, compare the parameters there between the working status (using ALSA API) and non-working status (OSS API). Takashi

1 0

[alsa-devel] USB audio class v2.0
by Daniel Mack 03 Apr '07

03 Apr '07

Hi, is there any efford taken to support devices conforming to the USB audio class in version 2.0? I'm just curious if anybody ever started work on that since I couldn't find any hint in the sources. Greets, Daniel

3 4

[alsa-devel] dereferencing pointer error in app but not in lib
by intellij 03 Apr '07

03 Apr '07

In the library, like pcm.c I can code like const char *snd_pcm_name(snd_pcm_t pcm) { return pcm->name; } At the same time, the same code in the application causes "dereferencing pointer to incomplete type" error. Note, the compiler does not tell that there is no such field in the pcm structure. It just thinks that the name has another type. Which one? thank you.

2 1

[alsa-devel] PCM low level - need help
by ODINOT Jean-Marc ETRALI 02 Apr '07

02 Apr '07

Hi Alsa gurus, I've to write an alsa driver for a custom card. I readed "writing an alsa driver" and other paper about alsa and drivers. Well I got a result, but not as good as I think it's possible with alsa. My card have 30 output slots (and 7 input slots), can be 8Khz or 16Khz (same rate for every slots), 16bits little endian by sample, can have one or more sample for each slot (same for every slots) in a frame, an hardware IT every frame (a frame is 30 slots :-). The purpose is to get, for each IT, a frame with all the datas. IT -> frame [channel1][channel2][channel3]...[channel30] channelX [sample1][sample2]...[sampleN] sampleX [16bitsle] At this time, I've something like a frame if I use channel30 with speaker-test -c30, but if I use channel1, alsa give me a buffer with many samples for channel1. And, just after oppening, I get an extra word. BTW I use .copy to get the buffer. My questions: -why the extra word (can I avoid it?)? -how set the buffer len to always have all my channels (used or not)? -is it possible to have more than a sample by channel contiguous? Think's it's all for today, but maybe your answers will give me more ideas :-) Thanx for replies (and please, forgive my awful english :-(, JM ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

2 1

Re: [alsa-devel] [Alsa-devel] [i8x0] docking station audio-out ceased to work
by Takashi Iwai 02 Apr '07

02 Apr '07

At Wed, 21 Mar 2007 23:57:32 +0100, Andreas Happe wrote: > > On Mon, 2007-03-19 at 18:53 +0100, Takashi Iwai wrote: > > Interesting. Usually AC97-SPSA specifies the AC97 slot for SPDIF > > output. The full (= 3) means the slot 10/11, 0 means slot 3/4. > > IIUC, AC97-SPSA must be 3 for docking station output working? > > If i remember correctly - but I was on holiday before the hardware > stopped to work (it was either set to 0 or 3, but I think that it was > 3). > > > If yes, try to set spdif_aclink=1 with the latest driver (HG > > version)... > > do you mean alsa-driver or alsa-kernel? I'm somehow reluctant to replace > my distribution's userspace utilities with self-compiled stuff (while i > would happily build a custom kernel). You need both trees to compile the driver. > BTW: a short link to a tutorial howto build hg kernels (or against what > kernel release). I checked it out, copied over the sound/ directory and > rebuild my 2.6.21-rc4 kernel. Copying to Linus kernel may work in most cases but sometimes not since alsa-kernel repo is updated for mm tree. The building HG driver is easy. Clone (both) repos, and run hgcompile script instead of configure. See alsa-driver/INSTALL. Takashi

2 10

[alsa-devel] how does hw:0, 1 eventually get mapped to /dev/snd/controlC0 and company? is this on the wiki somewhere?
by John Utz 02 Apr '07

02 Apr '07

how does this work? an explicit explanation would be great, but i'd even settle for a vague handwavy one at this point.

3 9

[alsa-devel] usleep() and nanosleep() timings seem inaccurate using ALSA
by Carlo Florendo 31 Mar '07

31 Mar '07

Good Day! After studying the intricacies of MIDI, I ended up writing an implementation of the MIDI protocol and file format. I then studied the ALSA sequencer API to be able to control a synthesizer keyboard and play MIDI files. I've used aplaymidi, aconnect, arecordmidi, and all those great ALSA utilities. They're very good! However, I wanted to have a simple ncurses based, command line MIDI sequencer meant for small Linux distributions such as DSL or Trustix so I began to write a command line sequencer using ALSA. I've encountered one problem about using usleep() and nanosleep() especially in 2.4 kernels. The core of my problem in my application is that the MIDI files slow down their tempo when pitch changes are made on a particular channel/track, while a NOTE-ON event is being played. For simplicity's sake, I'll give timing examples in terms of seconds and not bars nor measures. For example, if I've got a 2 second NOTE-ON playing a guitar note, all timings are ok. If the NOTE-ON is interspersed with CONTROLLER messages such as a pitch bend, the part which is supposed to sound in approximately 2 seconds, sounds much longer than that. The effect of this is that a ritardando is heard everytime short-duration CONTROLLER messages are sent. The problem disappears when I use the sequencer queues. However, the problem manifests itself when using direct output (i.e. not using queues, using snd_seq_event_output and snd_seq_drain_output one after the other). The reason why I would not want to use the queue is that I'd want the UI of my program (such as metronome ticks and counts) to somehow synch with the sound that is being heard. I've tried removing all UI elements, thinking that the UI controls might have caused the delay, but still the ritardando effect persists. The structure of my code is like this: parse_midi_file(); merge_all_tracks_to_one_single_track(); normalize_the_delta_times() for all events; do usleep(delta_time_for_each_event); play_the_event_using_alsa_direct_output(); done The ritardando effect is a bit lesser using 2.6 kernels. If I use queues, the problem disappears completely in 2.4 and 2.6 kernels. I don't hear the effect with simple MIDI files, such as with piano pieces. By the way, I intend to use my app with a 2.4 kernel. Does anybody have a solution to the delay problem? Please point out to me if I'm doing anything wrong since I've just been working with alsa-seq for a few months. Thank you very much. Best Regards, Carlo -- Carlo Florendo Softare Engineer/Network Co-Administrator Astra Philippines Inc. UP-Ayala Technopark, Diliman 1101, Quezon City Philippines http://www.astra.ph -- The Astra Group of Companies 5-3-11 Sekido, Tama City Tokyo 206-0011, Japan http://www.astra.co.jp

2 4

[alsa-devel] is statically linking libasound supported? because if i do it, i cant open the device!
by John Utz 31 Mar '07

31 Mar '07

i have a test app that works successfully with hw:0,1 on gnto if i link it dynamically: dmx-gnto jutz # ./atestminhw1 Playback device is hw:0,1 Stream parameters are 48000Hz, S16_LE, 6 channels Rate set to 48000Hz (requested 48000Hz) Buffer size range from 6 to 5461 Period size range from 3 to 2730 Using max buffer size 5460 Periods = 4 was set period_size = 1365 was set buffer_size = 5460 Channel: 0 Freq: 220.000000 but it fails to find hw:0,1 if i link it statically, is that expected behavior? dmx-gnto jutz # ./atestminhw1 Playback device is hw:0,1 Stream parameters are 48000Hz, S16_LE, 6 channels ALSA lib pcm.c:2109:(snd_pcm_open_conf) symbol _snd_pcm_hw_open is not defined inside (null) Failed: No such device or address this line succeeds with a dynamic linking and fails with if(0 > (iErr = snd_pcm_open(&pPcm, gs_pcNameDev, SND_PCM_STREAM_PLAYBACK, 0))) goto fin; here is my static link arguments: gcc -g -static -o atestminhw1 atestmin.c -lasound -lm -lpthread -ldl here is my dynamic link arguments: gcc -g -o atestminhw1 atestmin.c -lasound and, for completeness, here is my lame test app, derived from other test apps that come with the source: #include <stdio.h> #include <stdlib.h> #include <string.h> #include <sched.h> #include <errno.h> #include <getopt.h> #include <inttypes.h> #include <ctype.h> #include <byteswap.h> #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include <alsa/asoundlib.h> #include <sys/time.h> #include <math.h> #define MAX_CHANNELS 16 static char *gs_pcNameDev = "hw:0,1"; /* playback device */ //static char *gs_pcNameDev = "plughw"; /* playback device */ //static char *gs_pcNameDev = "default"; /* playback device */ static snd_pcm_format_t gs_SndPcmFmt = SND_PCM_FORMAT_S16; /* sample format */ static unsigned int gs_uRate = 48000; /* stream rate */ static unsigned int gs_uCntChan = 6; /* count of channels */ static unsigned int gs_uTimBuf = 0; /* ring buffer length in us */ static unsigned int gs_uTimPer = 0; /* period time in us */ static unsigned int gs_uCntPer = 4; /* number of periods */ static double gs_dFreq = 440; /* sinusoidal wave frequency in Hz */ static snd_pcm_uframes_t gs_uSizBuf; static snd_pcm_uframes_t gs_uSizPer; static int set_hwparams(snd_pcm_t *handle, snd_pcm_hw_params_t *params, snd_pcm_access_t access) { unsigned int uRate; int iErr; snd_pcm_uframes_t period_size_min; snd_pcm_uframes_t period_size_max; snd_pcm_uframes_t buffer_size_min; snd_pcm_uframes_t buffer_size_max; /* choose all parameters */ if (0 > (iErr = snd_pcm_hw_params_any(handle, params))) { printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(iErr)); return iErr; } /* set the interleaved read/write format */ if (0 > (iErr = snd_pcm_hw_params_set_access(handle, params, access))) { printf("Access type not available for playback: %s\n", snd_strerror(iErr)); return iErr; } /* set the sample format */ if (0 > (iErr = snd_pcm_hw_params_set_format(handle, params, gs_SndPcmFmt))) { printf("Sample format not available for playback: %s\n", snd_strerror(iErr)); return iErr; } /* set the count of channels */ if (0 > (iErr = snd_pcm_hw_params_set_channels(handle, params, gs_uCntChan))) { printf("Channels count (%i) not available for playbacks: %s\n", gs_uCntChan, snd_strerror(iErr)); return iErr; } /* set the stream rate */ uRate = gs_uRate; if (0 > (iErr = snd_pcm_hw_params_set_rate(handle, params, gs_uRate, 0))) { printf("Rate %iHz not available for playback: %s\n", gs_uRate, snd_strerror(iErr)); return iErr; } if (uRate != gs_uRate) { printf("Rate doesn't match (requested %iHz, get %iHz, err %d)\n", gs_uRate, uRate, iErr); return -EINVAL; } printf("Rate set to %iHz (requested %iHz)\n", uRate, gs_uRate); /* set the buffer time */ iErr = snd_pcm_hw_params_get_buffer_size_min(params, &buffer_size_min); iErr = snd_pcm_hw_params_get_buffer_size_max(params, &buffer_size_max); iErr = snd_pcm_hw_params_get_period_size_min(params, &period_size_min, NULL); iErr = snd_pcm_hw_params_get_period_size_max(params, &period_size_max, NULL); printf("Buffer size range from %lu to %lu\n",buffer_size_min, buffer_size_max); printf("Period size range from %lu to %lu\n",period_size_min, period_size_max); if (gs_uTimPer > 0) { printf("Requested period time %u us\n", gs_uTimPer); if (0 > (iErr = snd_pcm_hw_params_set_period_time_near(handle, params, &gs_uTimPer, NULL))) { printf("Unable to set period time %u us for playback: %s\n", gs_uTimPer, snd_strerror(iErr)); return iErr; } } if (gs_uTimBuf > 0) { printf("Requested buffer time %u us\n", gs_uTimBuf); if (0 > (iErr = snd_pcm_hw_params_set_buffer_time_near(handle, params, &gs_uTimBuf, NULL))) { printf("Unable to set buffer time %u us for playback: %s\n", gs_uTimBuf, snd_strerror(iErr)); return iErr; } } if (! gs_uTimBuf && ! gs_uTimPer) { gs_uSizBuf = buffer_size_max; if (! gs_uTimPer) gs_uSizBuf = (gs_uSizBuf / gs_uCntPer) * gs_uCntPer; printf("Using max buffer size %lu\n", gs_uSizBuf); if (0 > (iErr = snd_pcm_hw_params_set_buffer_size_near(handle, params, &gs_uSizBuf))) { printf("Unable to set buffer size %lu for playback: %s\n", gs_uSizBuf, snd_strerror(iErr)); return iErr; } } if (! gs_uTimBuf || ! gs_uTimPer) { printf("Periods = %u\n", gs_uCntPer); if (0 > (iErr = snd_pcm_hw_params_set_periods_near(handle, params, &gs_uCntPer, NULL))) { printf("Unable to set nperiods %u for playback: %s\n", gs_uCntPer, snd_strerror(iErr)); return iErr; } } snd_pcm_hw_params_get_buffer_size(params, &gs_uSizBuf); snd_pcm_hw_params_get_period_size(params, &gs_uSizPer, NULL); printf("was set period_size = %lu\n",gs_uSizPer); printf("was set buffer_size = %lu\n",gs_uSizBuf); if (2*gs_uSizPer > gs_uSizBuf) { printf("buffer to small, could not use\n"); return iErr; } /* write the parameters to device */ iErr = snd_pcm_hw_params(handle, params); if (iErr < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(iErr)); return iErr; } return 0; } static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams) { int iErr; /* get the current swparams */ if (0 > (iErr = snd_pcm_sw_params_current(handle, swparams))) { printf("Unable to determine current swparams for playback: %s\n", snd_strerror(iErr)); return iErr; } /* start the transfer when a buffer is full */ if (0 > (iErr = snd_pcm_sw_params_set_start_threshold(handle, swparams, gs_uSizBuf))) { printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(iErr)); return iErr; } /* allow the transfer when at least period_size frames can be processed */ if (0 > (iErr = snd_pcm_sw_params_set_avail_min(handle, swparams, gs_uSizPer))) { printf("Unable to set avail min for playback: %s\n", snd_strerror(iErr)); return iErr; } /* align all transfers to 1 sample */ if (0 > (iErr = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1))) { printf("Unable to set transfer align for playback: %s\n", snd_strerror(iErr)); return iErr; } /* write the parameters to the playback device */ if (0 > (iErr = snd_pcm_sw_params(handle, swparams))) { printf("Unable to set sw params for playback: %s\n", snd_strerror(iErr)); return iErr; } return 0; } static void generate_sine(uint8_t *pu8Frames, int iChannel, int iCntFrames, double *pdPhase, float fFreq) { double dPhase = *pdPhase; double dPhaseMax = 1.0 / fFreq; double dSizStep = 1.0 / (double)gs_uRate; double dRes; int iIdxChan; int32_t i32Res; int16_t *pi16Samp = (int16_t*) pu8Frames; while (iCntFrames-- > 0) { for(iIdxChan=0;iIdxChan<gs_uCntChan;iIdxChan++) if (iIdxChan==iChannel) { dRes = (sin((dPhase * 2 * M_PI) / dPhaseMax - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ i32Res = dRes; *pi16Samp++ = i32Res >>16; } else *pi16Samp++ = 0; dPhase += dSizStep; if (dPhase >= dPhaseMax) dPhase -= dPhaseMax; } *pdPhase = dPhase; } /* * Underrun and suspend recovery */ static int xrun_recovery(snd_pcm_t *pPcm, int iErr) { if (iErr == -EPIPE) { /* under-run */ iErr = snd_pcm_prepare(pPcm); if (iErr < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(iErr)); return 0; } else if (iErr == -ESTRPIPE) { while ((iErr = snd_pcm_resume(pPcm)) == -EAGAIN) sleep(1); /* wait until the suspend flag is released */ if (iErr < 0) { iErr = snd_pcm_prepare(pPcm); if (iErr < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(iErr)); } return 0; } return iErr; } static int write_loop(snd_pcm_t *pPcm, int iChan, int iNumPer, uint8_t *pu8Frames) { double phase = 0; int iErr, iIdxPer, iCntFrames; uint8_t *pu8FramesLocal; float fFreq = gs_dFreq*(iChan+1)/2; // provide an audible chan change cue printf("Channel: %d Freq: %f\n", iChan, fFreq); for(iIdxPer = 0; iIdxPer < iNumPer; iIdxPer++) { iCntFrames = gs_uSizPer; pu8FramesLocal = pu8Frames; generate_sine(pu8FramesLocal, iChan, gs_uSizPer, &phase, fFreq); while (0 < iCntFrames) { if (-EAGAIN == (iErr = snd_pcm_writei(pPcm, pu8FramesLocal, iCntFrames))) continue; if (0 > iErr) { printf("Write error: %d,%s\n", iErr, snd_strerror(iErr)); if (0 > xrun_recovery(pPcm, iErr)) { printf("xrun_recovery failed: %d,%s\n", iErr, snd_strerror(iErr)); return -1; } break; // skip one period } pu8FramesLocal += snd_pcm_frames_to_bytes(pPcm, iErr); iCntFrames -= iErr; } } if (gs_uSizBuf > iIdxPer * gs_uSizPer) { snd_pcm_drain(pPcm); snd_pcm_prepare(pPcm ); } return 0; } int main(int argc, char *argv[]) { int iErr; int iIdxChan; uint8_t *pu8Frames = NULL; snd_pcm_t *pPcm = NULL; snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; snd_pcm_hw_params_alloca(&hwparams); snd_pcm_sw_params_alloca(&swparams); printf("Playback device is %s\n", gs_pcNameDev); printf("Stream parameters are %iHz, %s, %i channels\n", gs_uRate, snd_pcm_format_name(gs_SndPcmFmt), gs_uCntChan); if(0 > (iErr = snd_pcm_open(&pPcm, gs_pcNameDev, SND_PCM_STREAM_PLAYBACK, 0))) goto fin; if(0 > (iErr = set_hwparams(pPcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED))) goto fin; if(0 > (iErr = set_swparams(pPcm, swparams))) goto fin; if(!(pu8Frames = malloc(snd_pcm_frames_to_bytes(pPcm, gs_uSizPer)))) goto fin; while(1) for(iIdxChan = 0; iIdxChan < gs_uCntChan; iIdxChan++) if (0> (iErr = write_loop(pPcm, iIdxChan, (gs_uRate*3)/gs_uSizPer, pu8Frames))) goto fin; iErr = EXIT_SUCCESS; fin: if(iErr) printf("Failed: %s\n",snd_strerror(iErr)); if(pu8Frames) free(pu8Frames); if(pPcm) snd_pcm_close(pPcm); exit(iErr); }

2 6

[alsa-devel] [2/4] 2.6.21-rc5: known regressions (v2)
by Adrian Bunk 30 Mar '07

30 Mar '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> Status : unknown Subject : forcedeth: sporadic under-load crashes References : http://lkml.org/lkml/2007/3/26/63 Submitter : Ingo Molnar <mingo(a)elte.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Ayaz Abdulla <aabdulla(a)nvidia.com> Status : problem is being debugged

1 0

[alsa-devel] 14rc3 fails to compile on vanilla 2.6.20.3
by Andrea Gozzi 30 Mar '07

30 Mar '07

Hi. I recently compiled a new kernel, 2.6.20.3, but when I tried to build alsa 14rc3 it failed. I also had troubles with the ati fglrx driver (missing declarations in headers - solved) so it could maybe be related to the new 20.x kernels. I attached logs for both the "configure" and "make". Andrea

2 3

[alsa-devel] Questions on Bluetooth Audio & Alsa Relationship
by Allyn, Mark A 30 Mar '07

30 Mar '07

Hello: Where is the correct place to discuss issues pertaining to snd-bt-sco and its relationship with Alsa? I am having trouble getting snd-bt-sco to work with Alsa 1.0.14rc3. I am getting a kernel oops for a NULL pointer somewhere in snd_bt_sco_playback_open. I looked on the blutooth-alsa sourceforge site, it seems that snd-bt-sco is no longer maintained and that they are going toward a solution using something called plugz. Plugz, however requires a kernel patch that works for kernels 2.6.19 and above, but not with 2.6.14, which is the kernel in the distro that I am using. I have attempted to ask some questions on the bluez developer email list on sourceforge, but I only got one response saying that the snd-bt-sco solution is more mature than the plugz development effort. I then asked about the kernel oops and have not got any response. What is the state of bluetooth audio development? Is it slacking off? Perhaps is the a plugz email list that I don't know about? Thank you for your help. Truly, Mark Allyn

2 1

[alsa-devel] Custom ioctl
by intellij 30 Mar '07

30 Mar '07

hi, how can I send custom request to my driver? Looking into the sources I suspect that sending ioctl() on opened hardware pcm will not reach the playback_ops.ioctl in my driver. thank you very much.

2 1

[alsa-devel] [PATCH] snd-usb-caiaq: Make playback work
by Karsten Wiese 29 Mar '07

29 Mar '07

Hi Takashi, some typo fixes. best regards, Karsten -----------------------------------------------------------------------> From: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Date: Thu, 29 Mar 2007 16:39:47 +0200 Subject: [PATCH] snd-usb-caiaq: Make playback work Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- sound/usb/caiaq/caiaq-audio.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index e80c8db..0414d76 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -388,8 +388,8 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] - = usb_buf[i]; + audio_buf[dev->audio_in_buf_pos[stream]++] = + usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) dev->audio_in_buf_pos[stream] = 0; @@ -436,19 +436,21 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, spin_lock(&dev->spinlock); for (i = 0; i < iso->length;) { - for (stream = 0; stream < dev->n_streams; stream++) { + for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - usb_buf[i++] - = audio_buf[dev->audio_out_buf_pos[stream]++]; + usb_buf[i] = + audio_buf[dev->audio_out_buf_pos[stream]]; + dev->period_out_count[stream]++; dev->audio_out_buf_pos[stream]++; if (dev->audio_out_buf_pos[stream] == sz) dev->audio_out_buf_pos[stream] = 0; } else - usb_buf[i++] = 0; + usb_buf[i] = 0; + } /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && @@ -456,7 +458,6 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); - } } spin_unlock(&dev->spinlock);

2 1

[alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 29 Mar '07

29 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hello, I'm looking for the maintainers of the Alsa OSS layer. It seems it is getting more seriously broken each month. The last serious updates were almost 7 months ago by tiwai and perex. Dev must have noticed the 'skips' on the end of an OSS stream. The complete broken big endian emulation and so on. What is currently happening in this area? Yours Sincerely, Stefan de Konink -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGByXUYH1+F2Rqwn0RCmU3AJ9O25z5wcCpXLDUnx61idhB7MGcYACcD0fh Fl95KN+x77vuC006U0mexJ8= =guiG -----END PGP SIGNATURE-----

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[alsa-devel] Function ap192_4wire_start in revo.c
by dustin＠seznam.cz 29 Mar '07

29 Mar '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O, I have come accross a bug in revo.c (used for major inspiration :) )The function static unsigned char ap192_4wire_start(struct snd_ice1712 *ice) returns value of current GPIO bits, which is unsigned int. However, return value of the function is unsigned char. This results in clearing all GPIO8-22 in next snd_ice1712_gpio_write call of the 4-wire protocol communication which is a little bit inconvenient :). Thanks for fixing in HG. Best regards, Pavel.

2 1

Re: [alsa-devel] Status of CMI9738 and SiS7012
by Takashi Iwai 29 Mar '07

29 Mar '07

Hi, Please don't drop Cc to alsa-devel... At Wed, 28 Mar 2007 10:23:22 -0600, Rob Franklin wrote: > > On 3/28/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 27 Mar 2007 10:44:23 -0600, > > Rob Franklin wrote: > > > > > > On 3/27/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Tue, 27 Mar 2007 08:49:05 -0600, > > > > Rob Franklin wrote: > > > > > > > > > > Hi All, > > > > > > > > > > I posted a question a couple of weeks ago to the alsa-users list > > > > > (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ > > > > > motherboard), but never got a response. > > > > > > > > > > I have been looking at the ALSA source code for the AC'97 codec on my > > > > > board (CMI9738) and the source code for the intel8x0 driver (for my > > > > > SiS 7012). However, as far as I can tell from looking at the source, > > > > > S/PDIF is not enabled/supported for this combination of devices. > > > > > Additionally, Googling around I haven't found any mention of people > > > > > getting spdif working with ALSA and a CM9738 / SiS7012 combo. > > > > > > > > > > I am wondering whether anyone on the developer list can confirm that > > > > > this is the case. Also, if this is the case can someone give me an > > > > > idea of what would be required to get this working. The system is > > > > > setup with dual boot and so I could grab some of the values from > > > > > windows using a PCI sniffer, and I might be able to do some hacking on > > > > > the source code and submit a patch if someone could help guide me in > > > > > the right direction. > > > > > > > > Are you sure it's CM9738? AFAIK, CM9738 doesn't support SPDIF at > > > > all, but CM9739 does... > > > > > > > That was my first thought too, but I pulled open the case last night > > > and verified that it was indeed a 9738. The download page from the > > > motherboard manufacturer has a driver for CMI9738 and a CMI9738/9739 > > > driver (http://www.ecsusa.com/downloads/drivers_sound.html). I am > > > wondering if there are two different versions of this chip. > > > > > > Here is the output from lspci -vv for the sound device: > > > > > > 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] > > > AC'97 Sound Controller (rev a0) > > > Subsystem: Elitegroup Computer Systems Unknown device 0a44 > > > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > > > ParErr- Stepping- SERR- FastB2B- > > > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > > > >TAbort- <TAbort- <MAbort- >SERR- <PERR- > > > Latency: 32 (13000ns min, 2750ns max) > > > Interrupt: pin C routed to IRQ 22 > > > Region 0: I/O ports at d000 [size=256] > > > Region 1: I/O ports at d400 [size=128] > > > Capabilities: <access denied> > > > > > > One thought that I had was to force the AC'97 driver to detect the > > > chip as a 9739 instead of a 9738 to see whether my 9738 can work as a > > > 9739. But I'm not sure how to do that short of modifying the source > > > code and recompiling the kernel module. > > > > First check whether its codec id is CMI9738, see > > /proc/asound/card0/codec97#0/* files. If it's really CM9738, you can > > hack pci/ac97/ac97_codec.c to replace patch_cm9738 with patch_cm9739 > > in snd_ac97_codec_ids[]. > > > Here are the first couple of lines from /proc/asound/card0/codec97#0/ac97#0-0 : > > 0-0/0: C-Media Electronics CMI9738 > > PCI Subsys Vendor: 0x1019 > PCI Subsys Device: 0x0a44 > > Last night I hacked the ac97_codec.c and recompiled snd-ac97-codec.ko, > and now the file reads > > 0-0/0: C-Media Electronics -- rob CMI9739 > > Everything seems to be working the same as it was before, and I can > get stereo sound from the line-out jack just fine. However, when I > try to play back sound using the spdif I get the following: > > $ aplay -D plug:iec958 ac3_48khz_diatonis_soal.ac3 > ALSA lib setup.c:555:(add_elem) Cannot obtain info for CTL elem > (MIXER,'IEC958 Playback Default',0,0,0): No such file or directory I guess the control element wasn't created indeed because of lack of SPDIF feature on the codec. patch_cm9739() checks the availability of SPDIF bit (SPCV) and turn off the whole SPDIF feature if the bit isn't set. You can avoid that check just for testing. If this doesn't work, the possible situation is that the SPDIF is transmitted directly from the controller chip without AC97 codec. Unfortunately, we have no hardware information about SIS7012, so don't know whether it's even possible, and how to control it. > Is there a script I need to run or file I need to update my list of > CTLs (I have seen references to alsaconf and snddev on the web but > neither seems to be included with my FC5 install of alsa-utils)? % alsactl -f somefile store > Also, is there any easy way to reload the snd-ac97-codec.ko module. I > tried removing the modules with "rmmod snd-ac97-codec" and "modprobe > snd-ac97-codec". The module came out ok, but I got a bunch of errors > like this when I tried to reinsert the modules: > > Mar 27 21:16:46 XXXX kernel: snd_ac97_codec: Unknown symbol snd_info_register > Mar 27 21:16:46 XXXX kernel: snd_ac97_codec: disagrees about version > of symbol snd_ctl_add > > So right now I am rebooting the machine every time I want to replace > snd-ac97-codec. Seems that you still have the old modules on memory while installing the new modules. First unload all snd related modules, then reload again. Otherwise, something wrong with the build with your kernel tree. I never used FC or RedHat, so can't answer about it. Takashi

2 2

[alsa-devel] Enabling SPDIF output on Turtle Beach Audio Advantage Micro (CMEDIA CM102S)
by Eric Sandeen 28 Mar '07

28 Mar '07

I have this Turtle Beach Audio Advantage Micro usb sound card, er, stick. Inside it has pretty much nothing but a C-Media CM102S chip. TB claims, as does C-Media, that it can do S/PDIF out, and I believe them based on the glowing red TOSLINK light. :) I found previous emails on this list (http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09979.html) suggesting that there is probably some magic to be applied to tell usb hardware to go into S/PDIF mode. I've captured some traces under windows, linked from http://sandeen.net/tbaam/index.html Specifically, http://sandeen.net/tbaam/2spkr-spdif.txt seems to be a fairly short burst of USB activity when I used the control panel to switch from 2 speaker mode to spdif mode. I see the exact same activity when switching from headphone mode to spdif mode. Strangely, switching from, say, spdif back to 2 speaker seems to generate a lot more traffic. The full annotated trace is here: http://sandeen.net/tbaam/usbsnoop1.txt In any case, are these traces helpful to anyone who understands how to talk to a C-Media CM102S usb sound card? Does it look possible to poke this hardware into S/PDIF mode with this information? Thanks, -Eric

1 0

[alsa-devel] how to subscribe to new maillist?
by Songmao Tian 28 Mar '07

28 Mar '07

Hello, I have tried several times on the web page http://mailman.alsa-project.org/mailman/listinfo/alsa-devel, but i can't receive the confirm mail, so how to subscribe? Tian

4 3

[alsa-devel] New code for conexant chipsets in -mm tree
by Rodrigo Luiz 28 Mar '07

28 Mar '07

Hi. I have a HDA NVidia, with a Conexant chipset CX20549 Venice (CXT5045) I tried to test the last alsa-kernel in -mm tree to support my mic. (2.6.21-rc4-mm1 and 2.6.21-rc5-mm2) When I try to load the hda-intel, I receive this message: Mar 27 11:29:08 dig kernel: hda_intel: azx_get_response timeout, switching to polling mode Mar 27 11:29:09 dig kernel: hda_intel: azx_get_response timeout, switching to single_cmd mode After loaded, the speakers doesn't work. Headphone works ok. Mic too. Look in code, I apply this comment and now the speakers works for me: --- patch_conexant.c.orig 2007-03-27 11:37:43.000000000 -0300 +++ patch_conexant.c 2007-03-27 11:38:02.000000000 -0300 @@ -822,7 +822,7 @@ case CXT5045_FUJITSU: spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; +// spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; But, I don't understand why, because this code is not executed for my chipset. Thanks, Rodrigo Luiz

3 8

[alsa-devel] snd-es1968 capture broken
by Rene Herman 28 Mar '07

28 Mar '07

Good day. Capture is broken on snd-es1968, at least if it is driving a ES1970MS-3D (ESS Canyon 3D). The captured stream is "stuttering" like mad. Kernel 2.6.20 (ALSA 1.0.14-rc1) with alsa-lib 1.0.11 (my distribution's version). Does not happen with a CS4236 in the same machine and same software version(s). Don't know if this driver has a specific maintainer. If needed I guess I can try to pin it down to some change. I hardly use capture on this card but it wasn't always broken... Rene.

2 15

Re: [alsa-devel] Basic DMA question
by Sven Johnsson 28 Mar '07

28 Mar '07

Rene, I have already read the ALSA driver document. What I really like to understand is when DMA transfers are invoked, and what the source and destination for the transfer is. Regards, Sven 2007/3/26, Rene Herman <rene.herman(a)gmail.com>: > On 03/26/2007 03:13 AM, Sven Johnsson wrote: > > > I'm new to writing ALSA drivers, and I could use some help on > > clarifying how DMA is used in ALSA. > > "Writing an ALSA Driver" is a useful document. Google will find it... > > By the way, the alsa-devel mailinglist recently moved from > sourceforge.net to alsa-project.org directly and the new list is > moderated for non-subscribers. I just approved your message from the > moderation queue manually, but please consider subscribing, at: > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Thanks, > Rene > > > > > > > > Regards, > > Sven > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > >

3 2

Re: [alsa-devel] Status of CMI9738 and SiS7012
by Takashi Iwai 28 Mar '07

28 Mar '07

At Tue, 27 Mar 2007 10:44:23 -0600, Rob Franklin wrote: > > On 3/27/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 27 Mar 2007 08:49:05 -0600, > > Rob Franklin wrote: > > > > > > Hi All, > > > > > > I posted a question a couple of weeks ago to the alsa-users list > > > (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ > > > motherboard), but never got a response. > > > > > > I have been looking at the ALSA source code for the AC'97 codec on my > > > board (CMI9738) and the source code for the intel8x0 driver (for my > > > SiS 7012). However, as far as I can tell from looking at the source, > > > S/PDIF is not enabled/supported for this combination of devices. > > > Additionally, Googling around I haven't found any mention of people > > > getting spdif working with ALSA and a CM9738 / SiS7012 combo. > > > > > > I am wondering whether anyone on the developer list can confirm that > > > this is the case. Also, if this is the case can someone give me an > > > idea of what would be required to get this working. The system is > > > setup with dual boot and so I could grab some of the values from > > > windows using a PCI sniffer, and I might be able to do some hacking on > > > the source code and submit a patch if someone could help guide me in > > > the right direction. > > > > Are you sure it's CM9738? AFAIK, CM9738 doesn't support SPDIF at > > all, but CM9739 does... > > > That was my first thought too, but I pulled open the case last night > and verified that it was indeed a 9738. The download page from the > motherboard manufacturer has a driver for CMI9738 and a CMI9738/9739 > driver (http://www.ecsusa.com/downloads/drivers_sound.html). I am > wondering if there are two different versions of this chip. > > Here is the output from lspci -vv for the sound device: > > 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] > AC'97 Sound Controller (rev a0) > Subsystem: Elitegroup Computer Systems Unknown device 0a44 > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Stepping- SERR- FastB2B- > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > >TAbort- <TAbort- <MAbort- >SERR- <PERR- > Latency: 32 (13000ns min, 2750ns max) > Interrupt: pin C routed to IRQ 22 > Region 0: I/O ports at d000 [size=256] > Region 1: I/O ports at d400 [size=128] > Capabilities: <access denied> > > One thought that I had was to force the AC'97 driver to detect the > chip as a 9739 instead of a 9738 to see whether my 9738 can work as a > 9739. But I'm not sure how to do that short of modifying the source > code and recompiling the kernel module. First check whether its codec id is CMI9738, see /proc/asound/card0/codec97#0/* files. If it's really CM9738, you can hack pci/ac97/ac97_codec.c to replace patch_cm9738 with patch_cm9739 in snd_ac97_codec_ids[]. Takashi

1 0

[alsa-devel] Status of CMI9738 and SiS7012
by Rob Franklin 27 Mar '07

27 Mar '07

Hi All, I posted a question a couple of weeks ago to the alsa-users list (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ motherboard), but never got a response. I have been looking at the ALSA source code for the AC'97 codec on my board (CMI9738) and the source code for the intel8x0 driver (for my SiS 7012). However, as far as I can tell from looking at the source, S/PDIF is not enabled/supported for this combination of devices. Additionally, Googling around I haven't found any mention of people getting spdif working with ALSA and a CM9738 / SiS7012 combo. I am wondering whether anyone on the developer list can confirm that this is the case. Also, if this is the case can someone give me an idea of what would be required to get this working. The system is setup with dual boot and so I could grab some of the values from windows using a PCI sniffer, and I might be able to do some hacking on the source code and submit a patch if someone could help guide me in the right direction. Thanks, Rob

2 1

[alsa-devel] Correct value for AK4114_DIF2 constant?
by dustin＠seznam.cz 27 Mar '07

27 Mar '07

Hello, While going through AK4114 support I have hit the constant AK4114_DIF2 in ak4114.h . I would say its value should be (1<<6) according to AK4114 datasheet. This is current situation: #define AK4114_DIF2 (1<<5) /* Audio Data Control */ #define AK4114_DIF1 (1<<5) /* Audio Data Control */ #define AK4114_DIF0 (1<<4) /* Audio Data Control */ I think cards using this code should experience related problems. E.g. in juli.c and revo.c the constant #define AK4114_DIF_I24I2S (AK4114_DIF2|AK4114_DIF1|AK4114_DIF0) /* STDO: I2S; LRCLK, BICK = Input */ is used, intended to set output serial data mode to I2S/slave perhaps compatible with ICE1712/24. This may be causing problems mentioned in revo.c: #if 0 /* FIXME: ak4114 makes the sound much lower due to some confliction, * so let's disable it right now... */ #define BUILD_AK4114_AP192 #endif I am trying to add support for SPDIF-IN of AudioTrak MI/ODI/O addon card (ice1724 + ak4114), hopefully this will be a little step forward. Thanks for any reaction. Best regards, Pavel.

2 1

[alsa-devel] capture line-out
by Tomas Carnecky 27 Mar '07

27 Mar '07

Is it possible to capture/record line-out (or simply 'what the user hears')? I'm writing 'fraps for linux' and so far I have video capturing implemented and that works great, but users are also requesting sound-capturing and I'm looking into how that could be done. thanks tom

2 10

[alsa-devel] [3/5] 2.6.21-rc5: known regressions
by Adrian Bunk 27 Mar '07

27 Mar '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 : Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> 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 : 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

1 0

Re: [alsa-devel] [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards
by Rask Ingemann Lambertsen 27 Mar '07

27 Mar '07

On Mon, Mar 12, 2007 at 12:28:00PM +0100, Takashi Iwai wrote: > At Mon, 12 Mar 2007 00:24:36 +0100, > Rask Ingemann Lambertsen wrote: > > > I tried to include u-law support, but it doesn't sound right and playback > > speed is about one third of what it should be. I could really use some > > documentation here. I'll remove it before submitting a final patch if I > > don't get it to work. > > OK. I never got u-law or a-law to work. I tried various ADPCM formats which gave correct playback speed, but much noise. There seems to be a Media Vision ADPCM format, which is probably what the Jazz16 wants, but I have no information about that format. FWIW, the playback_format/capture_format values I've tried are: #define SB_DSP_ALAW_OUTPUT_AUTO 0x1F #define SB_DSP_ALAW_INPUT_AUTO 0x2F #define SB_DSP_ULAW_OUTPUT_AUTO 0x7f #define SB_DSP_ULAW_INPUT_AUTO 0x8f #define SB_DSP_ADPCM_OUTPUT_AUTO 0x7C+1 #define SB_DSP_ADPCM_INPUT_AUTO 0x8C+1 > > Setting the irq of the MPU-401 port requires the SB part to be active and > > even worse, DSP commands need to be sent. Would it be better to just use the > > MPU-401 port without an irq and avoid the complexity? > > Isn't SB part already active? It's fine with non-irq version (which > uses timer) if the code gets too complicated, of course. It turned out that I can't disable just the MPU port. -- Rask Ingemann Lambertsen

1 0

[alsa-devel] [PATCH] PnP: Protocol for Media Vision Jazz16 sound cards
by Rask Ingemann Lambertsen 26 Mar '07

26 Mar '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add code to detect and PnP protocol to manage Media Vision Jazz16 sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch depends on the patch "ALSA: Support Media Vision Jazz16 chips" because the patched include/sound/sb.h is needed. drivers/pnp/Kconfig | 12 drivers/pnp/Makefile | 1 drivers/pnp/jazz16pnp.c | 555 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 568 insertions(+) --- linux-2.6.20/drivers/pnp/Makefile-orig 2007-02-19 23:24:22.000000000 +0100 +++ linux-2.6.20/drivers/pnp/Makefile 2007-02-19 23:24:22.000000000 +0100 @@ -8,3 +8,4 @@ obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPBIOS) += pnpbios/ obj-$(CONFIG_ISAPNP) += isapnp/ +obj-$(CONFIG_JAZZ16PNP) += jazz16pnp.o --- linux-2.6.20/drivers/pnp/jazz16pnp.c-orig 2007-03-11 22:11:58.000000000 +0100 +++ linux-2.6.20/drivers/pnp/jazz16pnp.c 2007-03-25 21:27:18.000000000 +0200 @@ -0,0 +1,555 @@ +/* + * Media Vision Jazz16 Plug & Play support. + * + * Copyright (c) 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * This driver manages two devices: + * + * Device 0: The game port. + * Device 1: The SoundBlaster look-alike part. + * + * Sources: + * sound/oss/sb_common.c + * sys/dev/isa/sbdsp.c (OpenBSD) + */ + +#include <linux/types.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16 Plug & Play support"); +MODULE_LICENSE("GPL"); + +int jazz16pnp_disable = false; /* Disable Jazz16 PnP. */ +module_param_named(disable, jazz16pnp_disable, bool, false); +MODULE_PARM_DESC(disable, "Jazz16 Plug & Play disable switch"); + +int jazz16pnp_verbose = false; /* Verbosity level. */ +module_param_named(verbose, jazz16pnp_verbose, bool, false); +MODULE_PARM_DESC(verbose, "Jazz16 Plug & Play verbosity level"); + +#define PFX "jazz16pnp: " +#define PFX_DEBUG KERN_DEBUG PFX +#define PFX_INFO KERN_INFO PFX +#define PFX_WARN KERN_WARNING PFX + +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_JAZZ16_SET_DMAINTR 0xFB + +extern void *pnp_alloc(long size); +extern void pnp_free_option(struct pnp_option *option); + +static struct pnp_dev *jazz16_gameport_dev = NULL; +static struct pnp_dev *jazz16_sb_dev = NULL; +static struct resource *jazz16pnp_fm_res = NULL; + +static uint jazz16_sb_port = 0, jazz16_sb_irq = 0; +static uint jazz16_sb_dma1 = 0, jazz16_sb_dma2 = 0; +static uint jazz16_mpu401_port = 0, jazz16_mpu401_irq = 0; + +static uint jazz16pnp_count = 0; + +static struct pnp_protocol jazz16pnp_protocol; + +static bool __init jazz16pnp_easy_option(struct pnp_dev *dev, uint pri, + struct pnp_port *port, uint nports, + struct pnp_irq *irq, uint nirqs, + struct pnp_dma *dma, uint ndmas) +{ + struct pnp_option *opt; + struct pnp_port *new_port; + struct pnp_irq *new_irq; + struct pnp_dma *new_dma; + uint i; + + switch (pri) { + case PNP_RES_PRIORITY_PREFERRED: + case PNP_RES_PRIORITY_ACCEPTABLE: + case PNP_RES_PRIORITY_FUNCTIONAL: + opt = pnp_register_dependent_option(dev, pri); + break; + + default: + opt = pnp_register_independent_option(dev); + break; + } + if (!opt) + return false; + + for (i = 0; i < nports; i ++) { + new_port = pnp_alloc(sizeof(*new_port)); + if (!new_port) + goto out_no_mem; + memcpy(new_port, &port[i], sizeof(*new_port)); + pnp_register_port_resource(opt, new_port); + } + for (i = 0; i < nirqs; i ++) { + new_irq = pnp_alloc(sizeof(*new_irq)); + if (!new_irq) + goto out_no_mem; + memcpy(new_irq, &irq[i], sizeof(*new_irq)); + pnp_register_irq_resource(opt, new_irq); + } + for (i = 0; i < ndmas; i ++) { + new_dma = pnp_alloc(sizeof(*new_dma)); + if (!new_dma) + goto out_no_mem; + memcpy(new_dma, &dma[i], sizeof(*new_dma)); + pnp_register_dma_resource(opt, new_dma); + } + return true; + +out_no_mem: + return false; +} + +static struct pnp_port jazz16pnp_game_ports[1] __initdata = { + { .min = SB_JAZZ16_CONFIG_PORT, + .max = SB_JAZZ16_CONFIG_PORT, + .size = 1, .align = 0, + .flags = PNP_PORT_FLAG_FIXED, }, +}; + +static int __init jazz16pnp_add_gameport(void) +{ + struct pnp_dev *dev; + struct pnp_id *dev_id; + + if (!(dev = pnp_alloc(sizeof(*dev)))) + goto out_no_dev; + if (!(dev_id = pnp_alloc(sizeof(*dev_id)))) + goto out_no_id; + if (!jazz16pnp_easy_option(dev, PNP_RES_PRIORITY_INVALID, + jazz16pnp_game_ports, 1, + NULL, 0, NULL, 0)) + goto out_no_option; + + dev->protocol = &jazz16pnp_protocol; + dev->capabilities = PNP_READ; + dev->active = 1; + dev->number = 0; + + pnp_init_resource_table(&dev->res); + dev->res.port_resource[0].start = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].flags = IORESOURCE_IO; + + memcpy(dev_id->id, "PNPb02f", PNP_ID_LEN); + pnp_add_id(dev_id, dev); + + jazz16_gameport_dev = dev; + pnp_add_device(dev); + return 0; + +out_no_option: + if (dev->independent) + pnp_free_option(dev->independent); + kfree(dev_id); +out_no_id: + kfree(dev); +out_no_dev: + return -ENOMEM; +}; + +static struct pnp_port jazz16pnp_ports[2] __initdata = { + { .min = 0x220, .max = 0x260, .align = 0x20, .size = 0x10 }, + { .min = 0x300, .max = 0x330, .align = 0x10, .size = 0x02 }, +}; + +static struct pnp_irq jazz16pnp_irqs[2] __initdata = { + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_dma jazz16pnp_dmas[2] __initdata = { + { .map = (1 << 1) | (1 << 3), .flags = IORESOURCE_DMA_8BIT, }, + { .map = (1 << 5) | (1 << 7), .flags = IORESOURCE_DMA_16BIT, }, +}; + +static int __init jazz16pnp_add_sb(void) +{ + struct pnp_dev *sbdev; + struct pnp_id *sbid; + + if (!(sbdev = pnp_alloc(sizeof(*sbdev)))) + goto out_no_sbdev; + if (!(sbid = pnp_alloc(sizeof(*sbid)))) + goto out_no_sbid; + + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_PREFERRED, + jazz16pnp_ports, 2, + jazz16pnp_irqs, 2, + jazz16pnp_dmas, 2)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 2, + jazz16pnp_irqs, 1, + jazz16pnp_dmas, 2)) + goto out_no_sboption; + memcpy(sbid->id, "PNPb00f", PNP_ID_LEN); + pnp_add_id(sbid, sbdev); + + pnp_init_resource_table(&sbdev->res); + sbdev->capabilities = PNP_READ | PNP_WRITE | PNP_CONFIGURABLE | PNP_DISABLE; + sbdev->protocol = &jazz16pnp_protocol; + sbdev->number = 1; + + jazz16_sb_dev = sbdev; + pnp_add_device(sbdev); + return 0; + +out_no_sboption: + if (sbdev->dependent) + pnp_free_option(sbdev->dependent); + kfree(sbid); +out_no_sbid: + kfree(sbdev); +out_no_sbdev: + return -ENOMEM; +} + +/* Configure the SB and MPU ports. Passing 0 for sbport disables the Jazz16. */ +static void jazz16_port_setup(const uint sbport, const uint mpuport) +{ + u8 config; + + switch (sbport) { + case 0x220: + case 0x240: + case 0x260: + config = sbport & 0x70; + jazz16_sb_port = sbport; + break; + + default: + config = 0; + jazz16_sb_port = 0; + break; + } + + switch (mpuport) { + case 0x300: + case 0x310: + case 0x320: + case 0x330: + config |= (mpuport >> 4) & 0x07; + jazz16_mpu401_port = mpuport; + break; + + default: + if (jazz16_sb_port) + BUG(); + } + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB port = 0x%x, MPU port = 0x%x.\n", + jazz16_sb_port, jazz16_mpu401_port); + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(config, SB_JAZZ16_CONFIG_PORT); +} + +static u8 jazz16_irq_config(uint irq) +{ + switch (irq) { + case 3: return 3; + case 5: return 1; + case 7: return 4; + case 9: return 2; + case 10: return 5; + case 15: return 6; + default: return 0; + } +} + +static u8 jazz16_dma_config(uint dma) +{ + switch (dma) { + case 1: return 1; + case 3: return 2; + case 5: return 3; + case 7: return 4; + default: return 0; + } +} + +/* The my_sbdsp_xxx functions were ripped from sound/isa/sb/sb_common.c. */ +#define BUSY_LOOPS 100000 + +static int my_sbdsp_command(unsigned int base, unsigned char val) +{ + int i; + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(base, STATUS)) & 0x80) == 0) { + outb(val, SBP1(base, COMMAND)); + return 1; + } + return 0; +} + +static int my_sbdsp_get_byte(unsigned int base) +{ + int val; + int i; + for (i = BUSY_LOOPS; i; i--) { + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + val = inb(SBP1(base, READ)); + return val; + } + } + return -ENODEV; +} + +static int my_sbdsp_reset(unsigned int base) +{ + int i; + + outb(1, SBP1(base, RESET)); + udelay(10); + outb(0, SBP1(base, RESET)); + udelay(30); + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + if (inb(SBP1(base, READ)) == 0xaa) + return 0; + else + break; + } + return -ENODEV; +} + +static int jazz16_irq_dma_setup(unsigned int base, uint sbirq, uint mpuirq, uint dma8, uint dma16) +{ + u8 dma_config, irq_config; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB irq %u, dma %u&%u, MPU irq %u.\n", + sbirq, dma8, dma16, mpuirq); + irq_config = jazz16_irq_config(sbirq) | jazz16_irq_config(mpuirq) << 4; + dma_config = jazz16_dma_config(dma8) | jazz16_dma_config(dma16) << 4; + if (!my_sbdsp_command(base, SB_JAZZ16_SET_DMAINTR)) + return 0; + + if (!my_sbdsp_command(base, dma_config)) + return 0; + jazz16_sb_dma1 = dma8; + jazz16_sb_dma2 = dma16; + + if (!my_sbdsp_command(base, irq_config)) + return 0; + jazz16_sb_irq = sbirq; + jazz16_mpu401_irq = mpuirq; + return 1; +} + +static int jazz16pnp_get_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + pnp_init_resource_table(res); + if (dev == jazz16_gameport_dev) { + res->port_resource[0].start = res->port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + res->port_resource[0].flags = IORESOURCE_IO; + } + else if (dev == jazz16_sb_dev) { + if (!jazz16_sb_port) + return 0; + res->port_resource[0].start = jazz16_sb_port; + res->port_resource[0].end = jazz16_sb_port + 0x10 - 1; + res->port_resource[0].flags = IORESOURCE_IO; + + res->irq_resource[0].start = res->irq_resource[0].end = jazz16_sb_irq; + res->irq_resource[0].flags = IORESOURCE_IRQ; + + res->dma_resource[0].start = res->dma_resource[0].end = jazz16_sb_dma1; + res->dma_resource[0].flags = IORESOURCE_DMA; + res->dma_resource[1].start = res->dma_resource[1].end = jazz16_sb_dma2; + res->dma_resource[1].flags = IORESOURCE_DMA; + + res->port_resource[1].start = jazz16_mpu401_port; + res->port_resource[1].end = jazz16_mpu401_port + 2 - 1; + res->port_resource[1].flags = IORESOURCE_IO; + + if (!jazz16_mpu401_irq) + return 0; + res->irq_resource[1].start = res->irq_resource[0].end = jazz16_mpu401_irq; + res->irq_resource[1].flags = IORESOURCE_IRQ; + } + else + return -EINVAL; + return 0; +} + +static int jazz16pnp_set_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + unsigned int sbirq, mpuirq, dma1, dma2; + + if (dev == jazz16_gameport_dev) + return -ENOSYS; + + else if (dev == jazz16_sb_dev) { + if ((res->port_resource[0].flags & (IORESOURCE_IO | IORESOURCE_UNSET)) == IORESOURCE_IO) + jazz16_port_setup(res->port_resource[0].start, jazz16_mpu401_port); + if ((res->port_resource[1].flags & (IORESOURCE_IO | IORESOURCE_UNSET)) == IORESOURCE_IO) + jazz16_port_setup(jazz16_sb_port, res->port_resource[1].start); + if ((res->dma_resource[0].flags & (IORESOURCE_DMA | IORESOURCE_UNSET)) == IORESOURCE_DMA) + dma1 = res->dma_resource[0].start; + else + dma1 = jazz16_sb_dma1; + if ((res->dma_resource[1].flags & (IORESOURCE_DMA | IORESOURCE_UNSET)) == IORESOURCE_DMA) + dma2 = res->dma_resource[1].start; + else + dma2 = jazz16_sb_dma2; + if ((res->irq_resource[0].flags & (IORESOURCE_IRQ | IORESOURCE_UNSET)) == IORESOURCE_IRQ) + sbirq = res->irq_resource[0].start; + else + sbirq = jazz16_sb_irq; + if ((res->irq_resource[1].flags & (IORESOURCE_IRQ | IORESOURCE_UNSET)) == IORESOURCE_IRQ) + mpuirq = res->irq_resource[1].start; + else + mpuirq = jazz16_mpu401_irq; + if (jazz16_sb_port) { + /* Block the OPL3 alias to avoid conflicts. */ + jazz16pnp_fm_res = request_region(0x388, 4, "SoundBlaster FM"); + jazz16_irq_dma_setup(jazz16_sb_port, sbirq, mpuirq, dma1, dma2); + } else + return -EBUSY; + dev->active = true; + return 0; + } + else + return -EINVAL; +} + +static int jazz16pnp_disable_resources(struct pnp_dev *dev) +{ + if (dev == jazz16_gameport_dev) { + return -ENOSYS; + } + else if (dev == jazz16_sb_dev) { + jazz16_irq_dma_setup(jazz16_sb_port, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + if (jazz16pnp_fm_res) { + release_resource(jazz16pnp_fm_res); + jazz16pnp_fm_res = NULL; + } + dev->active = false; + return 0; + } + else { + return -EINVAL; + } +} + +static struct pnp_protocol jazz16pnp_protocol = { + .name = "Jazz16 PnP", + .get = jazz16pnp_get_resources, + .set = jazz16pnp_set_resources, + .disable = jazz16pnp_disable_resources, +}; + +static uint __init jazz16_check(unsigned int base) +{ + if (my_sbdsp_reset(base)) + return 0; + if (!(my_sbdsp_command(base, SB_DSP_GET_JAZZ_VERSION))) + return 0; + return SB_VERSION_IS_JAZZ16(my_sbdsp_get_byte(base)); +} + +static int __init jazz16pnp_init(void) +{ + int err = 0; + unsigned int base, mpubase; + + if (jazz16pnp_disable == 1) { + printk(PFX_INFO "Jazz16 Plug & Play support disabled.\n"); + return 0; + } + if (!request_region(SB_JAZZ16_CONFIG_PORT, 1, "Jazz16 PnP probe")) + return -EBUSY; + + for (mpubase = 0x330; mpubase >= 0x300; mpubase -= 0x10) + if (request_region(mpubase, 2, "Jazz16 PnP probe")) + break; + if (mpubase < 0x300) + goto out_no_mpubase; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Scanning for Jazz16 cards...\n"); + + for (base = 0x260; base >= 0x220; base -= 0x20) { + if (!request_region(base, 16, "Jazz16 PnP probe")) + continue; + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Probing at 0x%x.\n", base); + jazz16_port_setup(base, mpubase); + if (jazz16_check(base)) + jazz16pnp_count++; + jazz16_irq_dma_setup(base, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + release_region(base, 16); + if (jazz16pnp_count) + break; + } + if (base < 0x220) + goto out_no_sbbase; + if (!jazz16pnp_count) { + err = -ENODEV; + goto out_no_devices; + } + if (pnp_register_protocol(&jazz16pnp_protocol) < 0) + err = -ENOMEM; + else { + if (jazz16pnp_add_gameport()) + printk(PFX_WARN "Failed to add game port.\n"); + if (jazz16pnp_add_sb()) + printk(PFX_WARN "Failed to add SB port.\n"); + err = 0; + } +out_no_devices: + printk(PFX_INFO "%u Jazz16 card%s detected.\n", jazz16pnp_count, + jazz16pnp_count == 1 ? "" : "s"); +out_no_sbbase: + release_region(mpubase, 2); +out_no_mpubase: + release_region(SB_JAZZ16_CONFIG_PORT, 1); + return err; +} +module_init(jazz16pnp_init); + +static void __exit jazz16pnp_exit(void) +{ + pnp_unregister_protocol(&jazz16pnp_protocol); +} +module_exit(jazz16pnp_exit); --- linux-2.6.20/drivers/pnp/Kconfig-orig 2007-02-19 23:23:26.000000000 +0100 +++ linux-2.6.20/drivers/pnp/Kconfig 2007-03-25 21:44:48.000000000 +0200 @@ -39,5 +39,17 @@ source "drivers/pnp/pnpacpi/Kconfig" source "drivers/pnp/pnpacpi/Kconfig" +# +# Jazz16 Plug and Play configuration +# +config JAZZ16PNP + bool "Jazz16 sound card Plug and Play support" + depends on PNP && ISA + help + Say Y here if you would like support for automatic configuration + of I/O ports, DMA channels and IRQ lines of Jazz16 sound cards. + + Say Y if you have such a card. Otherwise, say N. + endmenu -- Rask Ingemann Lambertsen

1 0

[alsa-devel] Can't subscribe to the list
by Ed Wright 26 Mar '07

26 Mar '07

Hi, I've tried 3 times over the last week or so to subscribe to this list with no success. I can't believe I'm doing anything wrong (how hard is it?), but I don't see any posts (in the archives) indicating any problems subscribing, so I'm at a loss. Any suggestions? Or can you subscribe me manually? Thanks, Ed

2 2

[alsa-devel] Support for Compal EL81 (Intel HDA/Realtek alc883)
by Carlo E. Prelz 26 Mar '07

26 Mar '07

Good evening. I am the lucky owner of a nice Compal EL80 laptop, sporting an Intel HDA audio controller with realtek alc883. Here is what I get with lspci -vvv: 00:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: COMPAL Electronics Inc Unknown device [14c0:0017] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- 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 22 Region 0: Memory at d2500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,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 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) I had very elementary operativity, but missed much, including for example microphone boost. I applied the following patch: --8<-----8<----8<----8<----8<----8<----8<----8<----8<----8<---8<-- --- /usr/src/alsa/alsa-kernel/pci/hda/patch_realtek.c 2007-02-15 20:00:04.000000000 +0100 +++ ./patch_realtek.c 2007-03-15 08:36:30.000000000 +0100 @@ -141,6 +141,7 @@ ALC883_ACER, ALC883_MEDION, ALC883_LAPTOP_EAPD, + ALC883_LAPTOP_COMPAL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5582,6 +5583,7 @@ [ALC883_ACER] = "acer", [ALC883_MEDION] = "medion", [ALC883_LAPTOP_EAPD] = "laptop-eapd", + [ALC883_LAPTOP_COMPAL] = "laptop-compal", [ALC883_AUTO] = "auto", }; @@ -5607,6 +5609,7 @@ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x14c0, 0x0017, "Compal EL80", ALC883_LAPTOP_COMPAL), {} }; @@ -5747,6 +5750,76 @@ .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, +#if 1 + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, +#else + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_fivestack_mixer, --- /usr/src/alsa/alsa-kernel/pci/hda/patch_realtek.c 2007-02-15 20:00:04.000000000 +0100 +++ ./patch_realtek.c 2007-03-15 08:36:30.000000000 +0100 @@ -141,6 +141,7 @@ ALC883_ACER, ALC883_MEDION, ALC883_LAPTOP_EAPD, + ALC883_LAPTOP_COMPAL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5582,6 +5583,7 @@ [ALC883_ACER] = "acer", [ALC883_MEDION] = "medion", [ALC883_LAPTOP_EAPD] = "laptop-eapd", + [ALC883_LAPTOP_COMPAL] = "laptop-compal", [ALC883_AUTO] = "auto", }; @@ -5607,6 +5609,7 @@ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x14c0, 0x0017, "Compal EL80", ALC883_LAPTOP_COMPAL), {} }; @@ -5747,6 +5750,76 @@ .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, +#if 1 + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, +#else + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_fivestack_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, + alc883_medion_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + }, + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_3ST_2ch_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + }, + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + }, + [ALC883_LAPTOP_COMPAL] = { + .mixers = { alc883_tagra_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc883_tagra_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_tagra_unsol_event, + .init_hook = alc883_tagra_automute, + }, +#endif }; --8<-----8<----8<----8<----8<----8<----8<----8<----8<----8<---8<-- As you can see, I did various tests with possible 'alc_config_preset' structures. I could only copy from other existing examples, since I have no idea how you would obtain the correct values to insert there. With the first example, the only one that is commented in (copied from LAPTOP_EAPD), I could obtain a working microphone boost. The thing is far from perfect, though. Suffice to say that I find the main volume in the cursor labeled 'Surround' in alsamixer. Is there any instruction about how to obtain the right information to configure this laptop of mine? Can you help? In this case, what other information may I provide? Thanks in advance Carlo - Den Haag, NL -- * Se la Strada e la sua Virtu' non fossero state messe da parte, * K * Carlo E. Prelz - fluido(a)fluido.as che bisogno ci sarebbe * di parlare tanto di amore e di rettitudine? (Chuang-Tzu)

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[alsa-devel] ftp.alsa-project.org
by Rene Herman 26 Mar '07

26 Mar '07

Hi Jaroslav. When you currently want to download a complete new release from ftp.alsa-project.org you have to scrounge around all the directories manually. Would it be easy to automate to have a /pub/alsa/<version>/ directory containing links to all the relevant packages? Ie: /pub/alsa/1.0.13/alsa-driver-1.0.13.tar.bz2 -> ../../driver/alsa-driver-1.13.tar.bz2 and same: alsa-lib-1.0.13.tar.bz2 -> ../../lib/alsa-lib-1.13.tar.bz2 alsa-plugins-1.0.13.tar.bz2 -> ../../plugins/alsa-plugins-1.13.tar.bz2 alsa-utils-1.0.13.tar.bz2 -> ../../utils/alsa-utils-1.13.tar.bz2 alsa-tools-1.0.13.tar.bz2 -> ../../tools/alsa-tools-1.13.tar.bz2 alsa-firmware-1.0.13.tar.bz2 -> .../firmware/alsa-firmware-1.13.tar.bz2 alsa-oss-lib-1.0.12.tar.bz2 -> ../../oss-lib/alsa-oss-lib-1.12.tar.bz2 with probably links /pub/alsa/stable -> 1.0.13 and /pub/alsa/devel -> 1.14rc3 and stuff like that. (by the way, as far as I'm concerned, the /pub/alsa locations wouldn't need to be links but could just as well be the main locations but if you want to keep the current layout in place...) Would make it a bit nicer to download a new release... Rene.

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[alsa-devel] Patch to p5vdc-x motherboard's sound
by sergiodaniel＠inf.ufg.br 26 Mar '07

26 Mar '07

This is a description to correct the problem of p5vdc-x motherboard's. In file hda_codec.c the line 1727 is: if (codec->bus->modelname && models) { the variable codec->bus->modelname not is reconized for asus pc5vdc-x motherboards. Your ship is "Analog Devices AD1986A" i correct this add the lines in the funcion static int patch_ad1986a(struct hda_codec *codec) . . . default: spec->num_mixers = 2; spec->mixers[1] = ad1986a_3st_mixers; spec->num_init_verbs = 3; spec->init_verbs[1] = ad1986a_3st_init_verbs; spec->init_verbs[2] = ad1986a_ch2_init; spec->channel_mode = ad1986a_modes; spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); spec->need_dac_fix = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; break; } The complete code is in www.inf.ufg.br/~sergiodaniel/alsa-driver-1.0.14_PATCH_rc3-p5vdc-x.tar.gz thank's.. Sérgio Daniel ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

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Re: [alsa-devel] [PATCH] azt3328.c: small cleanup patch
by Takashi Iwai 26 Mar '07

26 Mar '07

At Sun, 25 Mar 2007 21:06:56 +0200, Andreas Mohr wrote: > > Hi, > > - change "PCM" mixer control (pre/post 3D) to "PCM Output Route" > - improve snd_azf3328_debug_show_ports > - less aggressive module init message > - document Bass/Treble non-bug (prompted by user report - Thank You!!) > - add some items to card description > - add some I/O register documentation > - enhance copyright > > Patch done and hardware-tested on 2.6.21-rc4-mm1, hopefully applies cleanly > to current ALSA. > > Signed-off-by: Andreas Mohr <andi(a)lisas.de> Thanks, applied to HG tree now. (BTW, the address of alsa-devel ML was changed to alsa-devel(a)alsa-project.org) Takashi

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[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_a3d.c remove unused code
by Michal Piotrowski 26 Mar '07

26 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_a3d.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_a3d.c 2007-03-25 16:23:46.000000000 +0200 @@ -44,15 +44,6 @@ a3dsrc_SetTimeConsts(a3dsrc_t * a, short a3d_addrA(a->slice, a->source, A3D_A_CoeffTrackTC), CTrack); } -#if 0 -static void -a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack, - short *GTrack, short *CTrack) -{ - // stub! -} - -#endif /* Atmospheric absorbtion. */ static void @@ -95,41 +86,6 @@ a3dsrc_SetAtmosState(a3dsrc_t * a, short hwwrite(vortex->mmio, a3d_addrA(a->slice, a->source, A3D_A_y2), y2); } -#if 0 -static void -a3dsrc_GetAtmosTarget(a3dsrc_t * a, short *aa, short *b, short *c, - short *d, short *e) -{ -} -static void -a3dsrc_GetAtmosCurrent(a3dsrc_t * a, short *bb01, short *ab01, short *b2, - short *aa12, short *ba12) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *aa12 = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_A12Current)); - *ba12 = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_A12Current)); - *ab01 = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_B01Current)); - *bb01 = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_B01Current)); - *b2 = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_B2Current)); -} - -static void -a3dsrc_GetAtmosState(a3dsrc_t * a, short *x1, short *x2, short *y1, short *y2) -{ - -} - -#endif /* HRTF */ static void @@ -180,71 +136,6 @@ static void a3dsrc_SetHrtfOutput(a3dsrc_ a3d_addrA(a->slice, a->source, A3D_A_HrtfOutR), right); } -#if 0 -static void a3dsrc_GetHrtfTarget(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - - for (i = 0; i < HRTF_SZ; i++) - aa[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_HrtfTarget + (i << 2))); - for (i = 0; i < HRTF_SZ; i++) - b[i] = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, - A3D_B_HrtfTarget + (i << 2))); -} - -static void a3dsrc_GetHrtfCurrent(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - - for (i = 0; i < HRTF_SZ; i++) - aa[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_HrtfCurrent + (i << 2))); - for (i = 0; i < HRTF_SZ; i++) - b[i] = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, - A3D_B_HrtfCurrent + (i << 2))); -} - -static void a3dsrc_GetHrtfState(a3dsrc_t * a, a3d_Hrtf_t aa, a3d_Hrtf_t b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - // FIXME: verify this! - for (i = 0; i < HRTF_SZ; i++) - aa[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_HrtfDelayLine + (i << 2))); - for (i = 0; i < HRTF_SZ; i++) - b[i] = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, - A3D_B_HrtfDelayLine + (i << 2))); -} - -static void a3dsrc_GetHrtfOutput(a3dsrc_t * a, short *left, short *right) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *left = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_HrtfOutL)); - *right = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_HrtfOutR)); -} - -#endif - /* Interaural Time Difference. * "The other main clue that humans use to locate sounds, is called * Interaural Time Difference (ITD). The differences in distance from @@ -297,43 +188,6 @@ static void a3dsrc_SetItdDline(a3dsrc_t A3D_A_ITDDelayLine) + (i << 2), dline[i]); } -#if 0 -static void a3dsrc_GetItdTarget(a3dsrc_t * a, short *litd, short *ritd) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *ritd = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_ITDTarget)); - *litd = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_ITDTarget)); -} - -static void a3dsrc_GetItdCurrent(a3dsrc_t * a, short *litd, short *ritd) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - *ritd = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_ITDCurrent)); - *litd = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_ITDCurrent)); -} - -static void a3dsrc_GetItdDline(a3dsrc_t * a, a3d_ItdDline_t dline) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - int i; - - for (i = 0; i < DLINE_SZ; i++) - dline[i] = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, - A3D_A_ITDDelayLine + (i << 2))); -} - -#endif /* This is may be used for ILD Interaural Level Difference. */ static void a3dsrc_SetGainTarget(a3dsrc_t * a, short left, short right) @@ -352,37 +206,6 @@ static void a3dsrc_SetGainCurrent(a3dsrc (right << 0x10) | left); } -#if 0 -static void a3dsrc_GetGainTarget(a3dsrc_t * a, short *left, short *right) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *right = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_GainTarget)); - *left = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_GainTarget)); -} - -static void a3dsrc_GetGainCurrent(a3dsrc_t * a, short *left, short *right) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *right = - hwread(vortex->mmio, - a3d_addrA(a->slice, a->source, A3D_A_GainCurrent)); - *left = - hwread(vortex->mmio, - a3d_addrB(a->slice, a->source, A3D_B_GainCurrent)); -} - -/* CA3dIO this func seems to be inlined all over this place. */ -static void CA3dIO_WriteReg(a3dsrc_t * a, unsigned long addr, short aa, short b) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - hwwrite(vortex->mmio, addr, (aa << 0x10) | b); -} - -#endif /* Generic A3D stuff */ static void a3dsrc_SetA3DSampleRate(a3dsrc_t * a, int sr) @@ -422,28 +245,6 @@ static void a3dsrc_SetA3DPointerReg(a3ds hwwrite(vortex->mmio, A3D_SLICE_Pointers + ((a->slice) << 0xd), ptr); } -#if 0 -static void a3dsrc_GetA3DSampleRate(a3dsrc_t * a, int *sr) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *sr = ((hwread(vortex->mmio, A3D_SLICE_Control + (a->slice << 0xd)) - >> 3) & 0x1f); - //*sr = ((hwread(vortex->mmio, 0x19C38 + (this08<<0xd))>>3)&0x1f); -} - -static void a3dsrc_GetA3DControlReg(a3dsrc_t * a, unsigned long *ctrl) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *ctrl = hwread(vortex->mmio, A3D_SLICE_Control + ((a->slice) << 0xd)); -} - -static void a3dsrc_GetA3DPointerReg(a3dsrc_t * a, unsigned long *ptr) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - *ptr = hwread(vortex->mmio, A3D_SLICE_Pointers + ((a->slice) << 0xd)); -} - -#endif static void a3dsrc_ZeroSliceIO(a3dsrc_t * a) { vortex_t *vortex = (vortex_t *) (a->vortex); @@ -528,25 +329,6 @@ static void a3dsrc_ProgramPipe(a3dsrc_t //a3dsrc_SetHrtfTarget(a, A3dHrirSatTest, A3dHrirSatTest); } -/* VDB = Vortex audio Dataflow Bus */ -#if 0 -static void a3dsrc_ClearVDBData(a3dsrc_t * a, unsigned long aa) -{ - vortex_t *vortex = (vortex_t *) (a->vortex); - - // ((aa >> 2) << 8) - (aa >> 2) - hwwrite(vortex->mmio, - a3d_addrS(a->slice, A3D_SLICE_VDBDest) + (a->source << 2), 0); - hwwrite(vortex->mmio, - a3d_addrS(a->slice, - A3D_SLICE_VDBDest + 4) + (a->source << 2), 0); - /* - hwwrite(vortex->mmio, 0x19c00 + (((aa>>2)*255*4)+aa)*8, 0); - hwwrite(vortex->mmio, 0x19c04 + (((aa>>2)*255*4)+aa)*8, 0); - */ -} -#endif - /* A3D HwSource stuff. */ static void vortex_A3dSourceHw_Initialize(vortex_t * v, int source, int slice) @@ -622,7 +404,6 @@ static void vortex_Vort3D_connect(vortex return; #endif -#if 1 /* Alloc Xtalk mixin resources */ v->mixxtlk[0] = vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN); @@ -638,7 +419,6 @@ static void vortex_Vort3D_connect(vortex ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n"); return; } -#endif /* Connect A3D -> XTALK */ for (i = 0; i < 4; i++) { @@ -646,10 +426,7 @@ static void vortex_Vort3D_connect(vortex vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2), ADB_XTALKIN(i)); vortex_route(v, en, 0x11, ADB_A3DOUT(i * 2) + 1, ADB_XTALKIN(5 + i)); } -#if 0 - vortex_route(v, en, 0x11, ADB_XTALKOUT(0), ADB_EQIN(2)); - vortex_route(v, en, 0x11, ADB_XTALKOUT(1), ADB_EQIN(3)); -#else + /* Connect XTalk -> mixer */ vortex_route(v, en, 0x11, ADB_XTALKOUT(0), ADB_MIXIN(v->mixxtlk[0])); vortex_route(v, en, 0x11, ADB_XTALKOUT(1), ADB_MIXIN(v->mixxtlk[1])); @@ -671,7 +448,6 @@ static void vortex_Vort3D_connect(vortex v->mixxtlk[1], en ? MIX_DEFIGAIN : VOL_MIN); } -#endif } /* Initialize one single A3D source. */

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[alsa-devel] Basic DMA question
by Sven Johnsson 26 Mar '07

26 Mar '07

Hello all, I'm new to writing ALSA drivers, and I could use some help on clarifying how DMA is used in ALSA. Regards, Sven

1 0

[alsa-devel] hopeless about snd-hda-intel driver.
by Ustun ERGENOGLU 25 Mar '07

25 Mar '07

Hello everybody, for more than one month I have been crawling through the net to find a cure for my endless torment about ATI SB450 Realtek ALC861 soundcard. I have no problem with playback, but when it comes recording there is nothing wrong in the beginning but after sometime recording stops and I see a message about time out in dmesg output. I have made a bug report in the ALSA bug tracking system but got no response :(. It's here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2970 Can anybody, please, tell me where to begin to solve this problem. I can even start hacking ALSA code with my next to nothing knowledge about kernel driver programming as I have more than average c programming experience. This is more than enough motivation to start some coding, I suppose. I would greatly appreciate any recommendation or help. Thanks Ustun

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[alsa-devel] [PATCH -mm] sound/pci/trident/trident_synth.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/trident/trident_synth.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/trident/trident_synth.c 2007-03-25 00:41:40.000000000 +0100 @@ -772,19 +772,7 @@ static int snd_trident_synth_use(void *p voice->Vol = 0x3ff; voice->EC = 0x0fff; } -#if 0 - for (idx = 0; idx < info->midi_voices; idx++) { - port->midi_has_voices = 1; - voice = snd_trident_alloc_voice(trident, SNDRV_TRIDENT_VOICE_TYPE_MIDI, info->sender.client, info->sender.port); - if (voice == NULL) { - snd_trident_synth_free_voices(trident, info->sender.client, info->sender.port); - spin_unlock_irqrestore(&trident->reg_lock, flags); - return -EBUSY; - } - voice->Vol = 0x3ff; - voice->EC = 0x0fff; - } -#endif + spin_unlock_irqrestore(&trident->reg_lock, flags); return 0; }

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[alsa-devel] [PATCH -mm] sound/pci/trident/trident_main.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/trident/trident_main.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/trident/trident_main.c 2007-03-25 00:51:49.000000000 +0100 @@ -63,49 +63,6 @@ static int snd_trident_free(struct snd_t * common I/O routines */ - -#if 0 -static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) -{ - unsigned int val, tmp; - - printk("Trident voice %i:\n", voice); - outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR)); - val = inl(TRID_REG(trident, CH_LBA)); - printk("LBA: 0x%x\n", val); - val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); - printk("GVSel: %i\n", val >> 31); - printk("Pan: 0x%x\n", (val >> 24) & 0x7f); - printk("Vol: 0x%x\n", (val >> 16) & 0xff); - printk("CTRL: 0x%x\n", (val >> 12) & 0x0f); - printk("EC: 0x%x\n", val & 0x0fff); - if (trident->device != TRIDENT_DEVICE_ID_NX) { - val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS)); - printk("CSO: 0x%x\n", val >> 16); - printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff); - printk("FMS: 0x%x\n", val & 0x0f); - val = inl(TRID_REG(trident, CH_DX_ESO_DELTA)); - printk("ESO: 0x%x\n", val >> 16); - printk("Delta: 0x%x\n", val & 0xffff); - val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); - } else { // TRIDENT_DEVICE_ID_NX - val = inl(TRID_REG(trident, CH_NX_DELTA_CSO)); - tmp = (val >> 24) & 0xff; - printk("CSO: 0x%x\n", val & 0x00ffffff); - val = inl(TRID_REG(trident, CH_NX_DELTA_ESO)); - tmp |= (val >> 16) & 0xff00; - printk("Delta: 0x%x\n", tmp); - printk("ESO: 0x%x\n", val & 0x00ffffff); - val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL)); - printk("Alpha: 0x%x\n", val >> 20); - printk("FMS: 0x%x\n", (val >> 16) & 0x0f); - } - printk("FMC: 0x%x\n", (val >> 14) & 3); - printk("RVol: 0x%x\n", (val >> 7) & 0x7f); - printk("CVol: 0x%x\n", val & 0x7f); -} -#endif - /*--------------------------------------------------------------------------- unsigned short snd_trident_codec_read(struct snd_ac97 *ac97, unsigned short reg) @@ -495,15 +452,6 @@ void snd_trident_write_voice_regs(struct outl(regs[2], TRID_REG(trident, CH_START + 8)); outl(regs[3], TRID_REG(trident, CH_START + 12)); outl(regs[4], TRID_REG(trident, CH_START + 16)); - -#if 0 - printk("written %i channel:\n", voice->number); - printk(" regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0))); - printk(" regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4))); - printk(" regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8))); - printk(" regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12))); - printk(" regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16))); -#endif } EXPORT_SYMBOL(snd_trident_write_voice_regs); @@ -972,12 +920,7 @@ static int snd_trident_playback_prepare( voice->CVol = mix->cvol; voice->Pan = mix->pan; voice->Attribute = 0; -#if 0 - voice->Attribute = (1<<(30-16))|(2<<(26-16))| - (0<<(24-16))|(0x1f<<(19-16)); -#else voice->Attribute = 0; -#endif snd_trident_write_voice_regs(trident, voice); @@ -996,12 +939,7 @@ static int snd_trident_playback_prepare( evoice->Vol = 0x3ff; /* mute */ evoice->RVol = evoice->CVol = 0x7f; /* mute */ evoice->Pan = 0x7f; /* mute */ -#if 0 - evoice->Attribute = (1<<(30-16))|(2<<(26-16))| - (0<<(24-16))|(0x1f<<(19-16)); -#else evoice->Attribute = 0; -#endif snd_trident_write_voice_regs(trident, evoice); evoice->isync2 = 1; evoice->isync_mark = runtime->period_size; @@ -3804,14 +3742,7 @@ static irqreturn_t snd_trident_interrupt snd_trident_write_eso_reg(trident, voice, voice->ESO); snd_trident_start_voice(trident, voice->number); } -#if 0 - if (voice->extra) { - /* update CSO for extra voice to preserve sync */ - snd_trident_stop_voice(trident, voice->extra->number); - snd_trident_write_cso_reg(trident, voice->extra, 0); - snd_trident_start_voice(trident, voice->extra->number); - } -#endif + spin_unlock(&trident->reg_lock); snd_pcm_period_elapsed(voice->substream); spin_lock(&trident->reg_lock);

1 0

[alsa-devel] [PATCH -mm] sound/pci/pcxhr/pcxhr.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/pcxhr/pcxhr.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/pcxhr/pcxhr.c 2007-03-25 14:45:33.000000000 +0200 @@ -477,26 +477,6 @@ static int pcxhr_update_r_buffer(struct return err; } - -#if 0 -static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, snd_pcm_uframes_t *sample_count) -{ - struct pcxhr_rmh rmh; - int err; - pcxhr_t *chip = snd_pcm_substream_chip(stream->substream); - pcxhr_init_rmh(&rmh, CMD_PIPE_SAMPLE_COUNT); - pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, 0, 0, - 1<<stream->pipe->first_audio); - err = pcxhr_send_msg(chip->mgr, &rmh); - if (err == 0) { - *sample_count = ((snd_pcm_uframes_t)rmh.stat[0]) << 24; - *sample_count += (snd_pcm_uframes_t)rmh.stat[1]; - } - snd_printdd("PIPE_SAMPLE_COUNT = %lx\n", *sample_count); - return err; -} -#endif - static inline int pcxhr_stream_scheduled_get_pipe(struct pcxhr_stream *stream, struct pcxhr_pipe **pipe) {

1 0

[alsa-devel] [PATCH -mm] sound/pci/pcxhr/pcxhr_hwdep.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/pcxhr/pcxhr_hwdep.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/pcxhr/pcxhr_hwdep.c 2007-03-25 14:39:47.000000000 +0200 @@ -175,38 +175,6 @@ static int pcxhr_dsp_allocate_pipe( stru return 0; } -/* - * free playback/capture pipe (pcmp0/pcmc0) - */ -#if 0 -static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe) -{ - struct pcxhr_rmh rmh; - int capture_mask = 0; - int playback_mask = 0; - int err = 0; - - if (pipe->is_capture) - capture_mask = (1 << pipe->first_audio); - else - playback_mask = (1 << pipe->first_audio); - - /* stop one pipe */ - err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); - if (err < 0) - snd_printk(KERN_ERR "error stopping pipe!\n"); - /* release the pipe */ - pcxhr_init_rmh(&rmh, CMD_FREE_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, 0, 0); - err = pcxhr_send_msg(mgr, &rmh); - if (err < 0) - snd_printk(KERN_ERR "error pipe release (CMD_FREE_PIPE) err(%x)\n", err); - pipe->status = PCXHR_PIPE_UNDEFINED; - return err; -} -#endif - - static int pcxhr_config_pipes(struct pcxhr_mgr *mgr) { int err, i, j;

1 0

[alsa-devel] [PATCH -mm] sound/pci/ice1712/pontis.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/ice1712/pontis.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/ice1712/pontis.c 2007-03-25 00:56:33.000000000 +0100 @@ -767,12 +767,6 @@ static int __devinit pontis_init(struct WM_DAC_CTRL2, 0x0000, /* no deemphasis, no ZFLG */ WM_ADC_ATTEN_L, 0x0000, /* ADC muted */ WM_ADC_ATTEN_R, 0x0000, /* ADC muted */ -#if 0 - WM_ALC_CTRL1, 0x007b, /* */ - WM_ALC_CTRL2, 0x0000, /* */ - WM_ALC_CTRL3, 0x0000, /* */ - WM_NOISE_GATE, 0x0000, /* */ -#endif WM_DAC_MUTE, 0x0000, /* DAC unmute */ WM_ADC_MUX, 0x0003, /* ADC unmute, both CD/Line On */ };

1 0

[alsa-devel] [PATCH -mm] sound/pci/ice1712/prodigy192.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/ice1712/prodigy192.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/ice1712/prodigy192.c 2007-03-25 14:32:42.000000000 +0200 @@ -234,129 +234,6 @@ static int stac9460_adc_vol_put(struct s return change; } -#if 0 -/* - * Headphone Amplifier - */ -static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable) -{ - unsigned int tmp, tmp2; - - tmp2 = tmp = snd_ice1712_gpio_read(ice); - if (enable) - tmp |= AUREON_HP_SEL; - else - tmp &= ~ AUREON_HP_SEL; - if (tmp != tmp2) { - snd_ice1712_gpio_write(ice, tmp); - return 1; - } - return 0; -} - -static int aureon_get_headphone_amp(struct snd_ice1712 *ice) -{ - unsigned int tmp = snd_ice1712_gpio_read(ice); - - return ( tmp & AUREON_HP_SEL )!= 0; -} - -static int aureon_bool_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} - -static int aureon_hpamp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = aureon_get_headphone_amp(ice); - return 0; -} - - -static int aureon_hpamp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - return aureon_set_headphone_amp(ice,ucontrol->value.integer.value[0]); -} - -/* - * Deemphasis - */ -static int aureon_deemp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL2) & 0xf) == 0xf; - return 0; -} - -static int aureon_deemp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - int temp, temp2; - temp2 = temp = wm_get(ice, WM_DAC_CTRL2); - if (ucontrol->value.integer.value[0]) - temp |= 0xf; - else - temp &= ~0xf; - if (temp != temp2) { - wm_put(ice, WM_DAC_CTRL2, temp); - return 1; - } - return 0; -} - -/* - * ADC Oversampling - */ -static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[2] = { "128x", "64x" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = (wm_get(ice, WM_MASTER) & 0x8) == 0x8; - return 0; -} - -static int aureon_oversampling_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - int temp, temp2; - struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - - temp2 = temp = wm_get(ice, WM_MASTER); - - if (ucontrol->value.enumerated.item[0]) - temp |= 0x8; - else - temp &= ~0x8; - - if (temp != temp2) { - wm_put(ice, WM_MASTER, temp); - return 1; - } - return 0; -} -#endif - static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -424,36 +301,6 @@ static struct snd_kcontrol_new stac_cont .put = stac9460_adc_vol_put, .tlv = { .p = db_scale_adc } }, -#if 0 - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Route", - .info = wm_adc_mux_info, - .get = wm_adc_mux_get, - .put = wm_adc_mux_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Headphone Amplifier Switch", - .info = aureon_bool_info, - .get = aureon_hpamp_get, - .put = aureon_hpamp_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "DAC Deemphasis Switch", - .info = aureon_bool_info, - .get = aureon_deemp_get, - .put = aureon_deemp_put - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC Oversampling", - .info = aureon_oversampling_info, - .get = aureon_oversampling_get, - .put = aureon_oversampling_put - }, -#endif }; static int __devinit prodigy192_add_controls(struct snd_ice1712 *ice)

1 0

[alsa-devel] [PATCH -mm] sound/pci/ice1712/juli.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/ice1712/juli.c 2007-03-25 00:31:00.000000000 +0100 +++ linux-work3/sound/pci/ice1712/juli.c 2007-03-25 14:27:33.000000000 +0200 @@ -160,13 +160,6 @@ static int __devinit juli_init(struct sn int err; struct snd_akm4xxx *ak; -#if 0 - for (err = 0; err < 0x20; err++) - juli_ak4114_read(ice, err); - juli_ak4114_write(ice, 0, 0x0f); - juli_ak4114_read(ice, 0); - juli_ak4114_read(ice, 1); -#endif err = snd_ak4114_create(ice->card, juli_ak4114_read, juli_ak4114_write, @@ -175,14 +168,7 @@ static int __devinit juli_init(struct sn if (err < 0) return err; -#if 0 - /* it seems that the analog doughter board detection does not work - reliably, so force the analog flag; it should be very rare - to use Juli@ without the analog doughter board */ - ice->spec.juli.analog = (ice->gpio.get_data(ice) & GPIO_ANALOG_PRESENT) ? 0 : 1; -#else ice->spec.juli.analog = 1; -#endif if (ice->spec.juli.analog) { printk(KERN_INFO "juli@: analog I/O detected\n");

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_xtalk.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_xtalk.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_xtalk.c 2007-03-25 16:52:23.000000000 +0200 @@ -369,148 +369,6 @@ vortex_XtalkHw_SetRightXTStates(vortex_t hwwrite(vortex->mmio, 0x24534 + i * 0x24, arg_0[3]); } -#if 0 -static void -vortex_XtalkHw_GetLeftEQ(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24200 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24204 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24208 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x2420c + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x24210 + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24538) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x2453c) & 0xffff; -} - -static void -vortex_XtalkHw_GetRightEQ(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x242b4 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x242b8 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x242bc + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x242c0 + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x242c4 + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24540) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x24544) & 0xffff; -} - -static void -vortex_XtalkHw_GetLeftXT(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24368 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x2436C + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24370 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24374 + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x24378 + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24548) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x2454C) & 0xffff; -} - -static void -vortex_XtalkHw_GetRightXT(vortex_t * vortex, short *arg_0, short *arg_4, - xtalk_coefs_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x2441C + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24420 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24424 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24428 + i * 0x24); - coefs[i][4] = hwread(vortex->mmio, 0x2442C + i * 0x24); - } - *arg_0 = hwread(vortex->mmio, 0x24550) & 0xffff; - *arg_4 = hwread(vortex->mmio, 0x24554) & 0xffff; -} - -static void -vortex_XtalkHw_GetLeftEQStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24214 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24218 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x2421C + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24220 + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x244F8 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x244FC + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24500 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24504 + i * 0x24); -} - -static void -vortex_XtalkHw_GetRightEQStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x242C8 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x242CC + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x242D0 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x242D4 + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x24508 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2450C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24510 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24514 + i * 0x24); -} - -static void -vortex_XtalkHw_GetLeftXTStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x2437C + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24380 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24384 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x24388 + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x24518 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2451C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24520 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24524 + i * 0x24); -} - -static void -vortex_XtalkHw_GetRightXTStates(vortex_t * vortex, xtalk_instate_t arg_0, - xtalk_state_t coefs) -{ - int i; - - for (i = 0; i < 5; i++) { - coefs[i][0] = hwread(vortex->mmio, 0x24430 + i * 0x24); - coefs[i][1] = hwread(vortex->mmio, 0x24434 + i * 0x24); - coefs[i][2] = hwread(vortex->mmio, 0x24438 + i * 0x24); - coefs[i][3] = hwread(vortex->mmio, 0x2443C + i * 0x24); - } - arg_0[0] = hwread(vortex->mmio, 0x24528 + i * 0x24); - arg_0[1] = hwread(vortex->mmio, 0x2452C + i * 0x24); - arg_0[2] = hwread(vortex->mmio, 0x24530 + i * 0x24); - arg_0[3] = hwread(vortex->mmio, 0x24534 + i * 0x24); -} - -#endif /* Gains */ static void @@ -529,16 +387,6 @@ vortex_XtalkHw_SetGainsAllChan(vortex_t vortex_XtalkHw_SetGains(vortex, asXtalkGainsAllChan); } -#if 0 -static void vortex_XtalkHw_GetGains(vortex_t * vortex, xtalk_gains_t gains) -{ - int i; - - for (i = 0; i < XTGAINS_SZ; i++) - gains[i] = hwread(vortex->mmio, 0x244D0 + i * 4); -} - -#endif /* Delay parameters */ static void @@ -577,53 +425,6 @@ vortex_XtalkHw_SetRightDline(vortex_t * } } -#if 0 -static void -vortex_XtalkHw_GetDelay(vortex_t * vortex, unsigned short *right, - unsigned short *left) -{ - int esp0; - - esp0 = hwread(vortex->mmio, 0x24660); - *right = (esp0 >> 8) & 0x1f; - *left = (esp0 >> 0xd) & 0x1f; -} - -static void vortex_XtalkHw_GetLeftDline(vortex_t * vortex, xtalk_dline_t dline) -{ - int i; - - for (i = 0; i < 0x20; i++) { - dline[i] = - (hwread(vortex->mmio, 0x24000 + (i << 2)) & 0xffff) | - (hwread(vortex->mmio, 0x24080 + (i << 2)) << 0x10); - } -} - -static void vortex_XtalkHw_GetRightDline(vortex_t * vortex, xtalk_dline_t dline) -{ - int i; - - for (i = 0; i < 0x20; i++) { - dline[i] = - (hwread(vortex->mmio, 0x24100 + (i << 2)) & 0xffff) | - (hwread(vortex->mmio, 0x24180 + (i << 2)) << 0x10); - } -} - -#endif -/* Control/Global stuff */ - -#if 0 -static void vortex_XtalkHw_SetControlReg(vortex_t * vortex, u32 ctrl) -{ - hwwrite(vortex->mmio, 0x24660, ctrl); -} -static void vortex_XtalkHw_GetControlReg(vortex_t * vortex, u32 *ctrl) -{ - *ctrl = hwread(vortex->mmio, 0x24660); -} -#endif static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, u32 sr) { u32 temp; @@ -633,13 +434,6 @@ static void vortex_XtalkHw_SetSampleRate hwwrite(vortex->mmio, 0x24660, temp); } -#if 0 -static void vortex_XtalkHw_GetSampleRate(vortex_t * vortex, u32 *sr) -{ - *sr = (hwread(vortex->mmio, 0x24660) >> 3) & 0x1f; -} - -#endif static void vortex_XtalkHw_Enable(vortex_t * vortex) { u32 temp;

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_synth.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_synth.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_synth.c 2007-03-25 17:16:29.000000000 +0200 @@ -147,47 +147,6 @@ static void vortex_wt_connect(vortex_t * } } -/* Read WT Register */ -#if 0 -static int vortex_wt_GetReg(vortex_t * vortex, char reg, int wt) -{ - //int eax, esi; - - if (reg == 4) { - return hwread(vortex->mmio, WT_PARM(wt, 3)); - } - if (reg == 7) { - return hwread(vortex->mmio, WT_GMODE(wt)); - } - - return 0; -} - -/* WT hardware abstraction layer generic register interface. */ -static int -vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt, - u16 val) -{ - /* - int eax, edx; - - if (wt >= NR_WT) // 0x40 -> NR_WT - return 0; - - if ((reg - 0x20) > 0) { - if ((reg - 0x21) != 0) - return 0; - eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x208; // param 2 - } else { - eax = ((((b & 0xff) << 0xb) + (edx & 0xff)) << 4) + 0x20a; // param 3 - } - hwwrite(vortex->mmio, eax, c); - */ - return 1; -} - -/*public: static void __thiscall CWTHal::SetReg(unsigned char,int,unsigned long) */ -#endif static int vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, u32 val) @@ -315,81 +274,4 @@ static void vortex_wt_init(vortex_t * vo vortex_wt_SetReg(vortex, 0xa, edi, var10); /* ctrl */ } -/* Extract of CAdbTopology::SetVolume(struct _ASPVOLUME *) */ -#if 0 -static void vortex_wt_SetVolume(vortex_t * vortex, int wt, int vol[]) -{ - wt_voice_t *voice = &(vortex->wt_voice[wt]); - int ecx = vol[1], eax = vol[0]; - - /* This is pure guess */ - voice->parm0 &= 0xff00ffff; - voice->parm0 |= (vol[0] & 0xff) << 0x10; - voice->parm1 &= 0xff00ffff; - voice->parm1 |= (vol[1] & 0xff) << 0x10; - - /* This is real */ - hwwrite(vortex, WT_PARM(wt, 0), voice->parm0); - hwwrite(vortex, WT_PARM(wt, 1), voice->parm0); - - if (voice->this_1D0 & 4) { - eax >>= 8; - ecx = eax; - if (ecx < 0x80) - ecx = 0x7f; - voice->parm3 &= 0xFFFFC07F; - voice->parm3 |= (ecx & 0x7f) << 7; - voice->parm3 &= 0xFFFFFF80; - voice->parm3 |= (eax & 0x7f); - } else { - voice->parm3 &= 0xFFE03FFF; - voice->parm3 |= (eax & 0xFE00) << 5; - } - - hwwrite(vortex, WT_PARM(wt, 3), voice->parm3); -} - -/* Extract of CAdbTopology::SetFrequency(unsigned long arg_0) */ -static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr) -{ - wt_voice_t *voice = &(vortex->wt_voice[wt]); - u32 eax, edx; - - //FIXME: 64 bit operation. - eax = ((sr << 0xf) * 0x57619F1) & 0xffffffff; - edx = (((sr << 0xf) * 0x57619F1)) >> 0x20; - - edx >>= 0xa; - edx <<= 1; - if (edx) { - if (edx & 0x0FFF80000) - eax = 0x7fff; - else { - edx <<= 0xd; - eax = 7; - while ((edx & 0x80000000) == 0) { - edx <<= 1; - eax--; - if (eax == 0) ; - break; - } - if (eax) - edx <<= 1; - eax <<= 0xc; - edx >>= 0x14; - eax |= edx; - } - } else - eax = 0; - voice->parm0 &= 0xffff0001; - voice->parm0 |= (eax & 0x7fff) << 1; - voice->parm1 = voice->parm0 | 1; - // Wt: this_1D4 - //AuWt::WriteReg((ulong)(this_1DC<<4)+0x200, (ulong)this_1E4); - //AuWt::WriteReg((ulong)(this_1DC<<4)+0x204, (ulong)this_1E8); - hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0); - hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1); -} -#endif - /* End of File */

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_wt.h remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> mjander(a)users.sourceforge.org --- linux-work5/sound/pci/au88x0/au88x0_wt.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_wt.h 2007-03-25 16:57:17.000000000 +0200 @@ -33,25 +33,6 @@ #define WT_PARM(x,y) (((WT_BAR(x))+ 0x80 +(((x)&0x1f)<<2)+(y))<<2) /* 0x0200 */ #define WT_DELAY(x,y) (((WT_BAR(x))+ 0x100 +(((x)&0x1f)<<2)+(y))<<2) /* 0x0400 */ -/* Numeric indexes used by SetReg() and GetReg() */ -#if 0 -enum { - run = 0, /* 0 W 1:run 0:stop */ - parm0, /* 1 W filter, samplerate */ - parm1, /* 2 W filter, samplerate */ - parm2, /* 3 W */ - parm3, /* 4 RW volume. This value is calculated using floating point ops. */ - sramp, /* 5 W */ - mute, /* 6 W 1:mute, 0:unmute */ - gmode, /* 7 RO Looks like only bit0 is used. */ - aramp, /* 8 W */ - mramp, /* 9 W */ - ctrl, /* a W */ - delay, /* b W All 4 values are written at once with same value. */ - dsreg, /* c (R)W */ -} wt_reg; -#endif - typedef struct { u32 parm0; /* this_1E4 */ u32 parm1; /* this_1E8 */

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0.h remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0.h 2007-03-25 16:32:57.000000000 +0200 @@ -253,11 +253,6 @@ static void vortex_wt_init(vortex_t * vo static void vortex_route(vortex_t * vortex, int en, unsigned char channel, unsigned char source, unsigned char dest); -#if 0 -static void vortex_routes(vortex_t * vortex, int en, unsigned char channel, - unsigned char source, unsigned char dest0, - unsigned char dest1); -#endif static void vortex_connection_mixin_mix(vortex_t * vortex, int en, unsigned char mixin, unsigned char mix, int a);

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_core.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_core.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_core.c 2007-03-25 17:06:31.000000000 +0200 @@ -97,85 +97,6 @@ static void vortex_mixer_dis_sr(vortex_t hwread(vortex->mmio, VORTEX_MIXER_SR) & ~(0x1 << channel)); } -#if 0 -static void -vortex_mix_muteinputgain(vortex_t * vortex, unsigned char mix, - unsigned char channel) -{ - hwwrite(vortex->mmio, VORTEX_MIX_INVOL_A + ((mix << 5) + channel), - 0x80); - hwwrite(vortex->mmio, VORTEX_MIX_INVOL_B + ((mix << 5) + channel), - 0x80); -} - -static int vortex_mix_getvolume(vortex_t * vortex, unsigned char mix) -{ - int a; - a = hwread(vortex->mmio, VORTEX_MIX_VOL_A + (mix << 2)) & 0xff; - //FP2LinearFrac(a); - return (a); -} - -static int -vortex_mix_getinputvolume(vortex_t * vortex, unsigned char mix, - int channel, int *vol) -{ - int a; - if (!(mchannels[mix] & (1 << channel))) - return 0; - a = hwread(vortex->mmio, - VORTEX_MIX_INVOL_A + (((mix << 5) + channel) << 2)); - /* - if (rampchs[mix] == 0) - a = FP2LinearFrac(a); - else - a = FP2LinearFracWT(a); - */ - *vol = a; - return (0); -} - -static unsigned int vortex_mix_boost6db(unsigned char vol) -{ - return (vol + 8); /* WOW! what a complex function! */ -} - -static void vortex_mix_rampvolume(vortex_t * vortex, int mix) -{ - int ch; - char a; - // This function is intended for ramping down only (see vortex_disableinput()). - for (ch = 0; ch < 0x20; ch++) { - if (((1 << ch) & rampchs[mix]) == 0) - continue; - a = hwread(vortex->mmio, - VORTEX_MIX_INVOL_B + (((mix << 5) + ch) << 2)); - if (a > -126) { - a -= 2; - hwwrite(vortex->mmio, - VORTEX_MIX_INVOL_A + - (((mix << 5) + ch) << 2), a); - hwwrite(vortex->mmio, - VORTEX_MIX_INVOL_B + - (((mix << 5) + ch) << 2), a); - } else - vortex_mix_killinput(vortex, mix, ch); - } -} - -static int -vortex_mix_getenablebit(vortex_t * vortex, unsigned char mix, int mixin) -{ - int addr, temp; - if (mixin >= 0) - addr = mixin; - else - addr = mixin + 3; - addr = ((mix << 3) + (addr >> 2)) << 2; - temp = hwread(vortex->mmio, VORTEX_MIX_ENIN + addr); - return ((temp >> (mixin & 3)) & 1); -} -#endif static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, unsigned char vol) @@ -500,61 +421,6 @@ vortex_src_persist_convratio(vortex_t * return temp; } -#if 0 -static void vortex_src_slowlock(vortex_t * vortex, unsigned char src) -{ - int temp; - - hwwrite(vortex->mmio, VORTEX_SRC_DRIFT2 + (src << 2), 1); - hwwrite(vortex->mmio, VORTEX_SRC_DRIFT0 + (src << 2), 0); - temp = hwread(vortex->mmio, VORTEX_SRC_U0 + (src << 2)); - if (temp & 0x200) - hwwrite(vortex->mmio, VORTEX_SRC_U0 + (src << 2), - temp & ~0x200L); -} - -static void -vortex_src_change_convratio(vortex_t * vortex, unsigned char src, int ratio) -{ - int temp, a; - - if ((ratio & 0x10000) && (ratio != 0x10000)) { - if (ratio & 0x3fff) - a = (0x11 - ((ratio >> 0xe) & 0x3)) - 1; - else - a = (0x11 - ((ratio >> 0xe) & 0x3)) - 2; - } else - a = 0xc; - temp = hwread(vortex->mmio, VORTEX_SRC_U0 + (src << 2)); - if (((temp >> 4) & 0xf) != a) - hwwrite(vortex->mmio, VORTEX_SRC_U0 + (src << 2), - (temp & 0xf) | ((a & 0xf) << 4)); - - vortex_src_persist_convratio(vortex, src, ratio); -} - -static int -vortex_src_checkratio(vortex_t * vortex, unsigned char src, - unsigned int desired_ratio) -{ - int hw_ratio, lifeboat = 0; - - hw_ratio = hwread(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2)); - - while (hw_ratio != desired_ratio) { - hwwrite(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2), desired_ratio); - - if ((lifeboat++) > 15) { - printk(KERN_ERR "Vortex: could not set src-%d from %d to %d\n", - src, hw_ratio, desired_ratio); - break; - } - } - - return hw_ratio; -} - -#endif /* Objective: Set samplerate for given SRC module. Arguments: @@ -784,19 +650,6 @@ vortex_fifo_clearadbdata(vortex_t * vort (((fifo << FIFO_SIZE_BITS) + x) << 2), 0); } -#if 0 -static void vortex_fifo_adbinitialize(vortex_t * vortex, int fifo, int j) -{ - vortex_fifo_clearadbdata(vortex, fifo, FIFO_SIZE); -#ifdef CHIP_AU8820 - hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), - (FIFO_U1 | ((j & FIFO_MASK) << 0xb))); -#else - hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), - (FIFO_U1 | ((j & FIFO_MASK) << 0xc))); -#endif -} -#endif static void vortex_fifo_setadbvalid(vortex_t * vortex, int fifo, int en) { hwwrite(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2), @@ -1329,22 +1182,6 @@ static void vortex_adbdma_pausefifo(vort dma->fifo_status = FIFO_PAUSE; } -#if 0 // Using pause instead -static void vortex_adbdma_stopfifo(vortex_t * vortex, int adbdma) -{ - stream_t *dma = &vortex->dma_adb[adbdma]; - - int this_4 = 0, this_8 = 0; - if (dma->fifo_status == FIFO_START) - vortex_fifo_setadbctrl(vortex, adbdma, dma->dma_unknown, - this_4, this_8, 0, 0); - else if (dma->fifo_status == FIFO_STOP) - return; - dma->fifo_status = FIFO_STOP; - dma->fifo_enabled = 0; -} - -#endif /* WTDMA */ #ifndef CHIP_AU8810 @@ -1483,22 +1320,6 @@ static int vortex_wtdma_bufshift(vortex_ return delta; } -#if 0 -static void -vortex_wtdma_getposition(vortex_t * vortex, int wtdma, int *subbuf, int *pos) -{ - int temp; - temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)); - *subbuf = (temp >> WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK; - *pos = temp & POS_MASK; -} - -static int vortex_wtdma_getcursubuffer(vortex_t * vortex, int wtdma) -{ - return ((hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2)) >> - POS_SHIFT) & POS_MASK); -} -#endif static int inline vortex_wtdma_getlinearpos(vortex_t * vortex, int wtdma) { stream_t *dma = &vortex->dma_wt[wtdma]; @@ -1743,40 +1564,6 @@ vortex_route(vortex_t * vortex, int en, } } -#if 0 -static void -vortex_routes(vortex_t * vortex, int en, unsigned char channel, - unsigned char source, unsigned char dest0, unsigned char dest1) -{ - ADBRamLink route[2]; - - route[0] = ((source & ADB_MASK) << ADB_SHIFT) | (dest0 & ADB_MASK); - route[1] = ((source & ADB_MASK) << ADB_SHIFT) | (dest1 & ADB_MASK); - - if (en) { - vortex_adb_addroutes(vortex, channel, route, 2); - if ((source < (OFFSET_SRCOUT + NR_SRC)) - && (source >= (OFFSET_SRCOUT))) - vortex_src_addWTD(vortex, (source - OFFSET_SRCOUT), - channel); - else if ((source < (OFFSET_MIXOUT + NR_MIXOUT)) - && (source >= (OFFSET_MIXOUT))) - vortex_mixer_addWTD(vortex, - (source - OFFSET_MIXOUT), channel); - } else { - vortex_adb_delroutes(vortex, channel, route[0], route[1]); - if ((source < (OFFSET_SRCOUT + NR_SRC)) - && (source >= (OFFSET_SRCOUT))) - vortex_src_delWTD(vortex, (source - OFFSET_SRCOUT), - channel); - else if ((source < (OFFSET_MIXOUT + NR_MIXOUT)) - && (source >= (OFFSET_MIXOUT))) - vortex_mixer_delWTD(vortex, - (source - OFFSET_MIXOUT), channel); - } -} - -#endif /* Route two sources to same target. Sources must be of same class !!! */ static void vortex_routeLRT(vortex_t * vortex, int en, unsigned char ch, @@ -1899,44 +1686,6 @@ vortex_connection_mix_src(vortex_t * vor vortex_mix_setvolumebyte(vortex, mix, MIX_DEFOGAIN); // added to original code. } -#if 0 -static void -vortex_connection_adbdma_src_src(vortex_t * vortex, int en, - unsigned char channel, - unsigned char adbdma, unsigned char src0, - unsigned char src1) -{ - vortex_routes(vortex, en, channel, ADB_DMA(adbdma), - ADB_SRCIN(src0), ADB_SRCIN(src1)); -} - -// Connect two mix to AdbDma. -static void -vortex_connection_mix_mix_adbdma(vortex_t * vortex, int en, - unsigned char ch, unsigned char mix0, - unsigned char mix1, unsigned char adbdma) -{ - - ADBRamLink routes[2]; - routes[0] = - (((mix0 + - OFFSET_MIXOUT) & ADB_MASK) << ADB_SHIFT) | (adbdma & ADB_MASK); - routes[1] = - (((mix1 + OFFSET_MIXOUT) & ADB_MASK) << ADB_SHIFT) | ((adbdma + - 0x20) & - ADB_MASK); - if (en) { - vortex_adb_addroutes(vortex, ch, routes, 0x2); - vortex_mixer_addWTD(vortex, mix0, ch); - vortex_mixer_addWTD(vortex, mix1, ch); - } else { - vortex_adb_delroutes(vortex, ch, routes[0], routes[1]); - vortex_mixer_delWTD(vortex, mix0, ch); - vortex_mixer_delWTD(vortex, mix1, ch); - } -} -#endif - /* CODEC connect. */ static void @@ -1946,7 +1695,6 @@ vortex_connect_codecplay(vortex_t * vort vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1)); #else -#if 1 // Connect front channels through EQ. vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_EQIN(0)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_EQIN(1)); @@ -1965,11 +1713,6 @@ vortex_connect_codecplay(vortex_t * vort ADB_CODECOUT(1 + 4)); //printk("SDAC detected "); } -#else - // Use plain direct output to codec. - vortex_connection_mix_adb(vortex, en, 0x11, mixers[0], ADB_CODECOUT(0)); - vortex_connection_mix_adb(vortex, en, 0x11, mixers[1], ADB_CODECOUT(1)); -#endif #endif } @@ -2356,20 +2099,6 @@ static void vortex_settimer(vortex_t * v hwwrite(vortex->mmio, VORTEX_IRQ_STAT, period); } -#if 0 -static void vortex_enable_timer_int(vortex_t * card) -{ - hwwrite(card->mmio, VORTEX_IRQ_CTRL, - hwread(card->mmio, VORTEX_IRQ_CTRL) | IRQ_TIMER | 0x60); -} - -static void vortex_disable_timer_int(vortex_t * card) -{ - hwwrite(card->mmio, VORTEX_IRQ_CTRL, - hwread(card->mmio, VORTEX_IRQ_CTRL) & ~IRQ_TIMER); -} - -#endif static void vortex_enable_int(vortex_t * card) { // CAsp4ISR__EnableVortexInt_void_

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0_eq.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0_eq.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0_eq.c 2007-03-25 17:13:31.000000000 +0200 @@ -142,34 +142,6 @@ static void vortex_EqHw_SetRightStates(v } } -#if 0 -static void vortex_EqHw_GetTimeConsts(vortex_t * vortex, u16 * a, u16 * b) -{ - *a = hwread(vortex->mmio, 0x2b3c4); - *b = hwread(vortex->mmio, 0x2b3c8); -} - -static void vortex_EqHw_GetLeftCoefs(vortex_t * vortex, u16 a[]) -{ - -} - -static void vortex_EqHw_GetRightCoefs(vortex_t * vortex, u16 a[]) -{ - -} - -static void vortex_EqHw_GetLeftStates(vortex_t * vortex, u16 * a, u16 b[]) -{ - -} - -static void vortex_EqHw_GetRightStates(vortex_t * vortex, u16 * a, u16 b[]) -{ - -} - -#endif /* Mix Gains */ static void vortex_EqHw_SetBypassGain(vortex_t * vortex, u16 a, u16 b) { @@ -190,22 +162,6 @@ static void vortex_EqHw_SetA3DBypassGain hwwrite(vortex->mmio, 0x2b3f8, b); } -#if 0 -static void vortex_EqHw_SetCurrBypassGain(vortex_t * vortex, u16 a, u16 b) -{ - - hwwrite(vortex->mmio, 0x2b3d0, a); - hwwrite(vortex->mmio, 0x2b3e8, b); -} - -static void vortex_EqHw_SetCurrA3DBypassGain(vortex_t * vortex, u16 a, u16 b) -{ - - hwwrite(vortex->mmio, 0x2b3dc, a); - hwwrite(vortex->mmio, 0x2b3f4, b); -} - -#endif static void vortex_EqHw_SetLeftGainsSingleTarget(vortex_t * vortex, u16 index, u16 b) { @@ -258,68 +214,6 @@ static void vortex_EqHw_SetRightGainsCur } } -#if 0 -static void vortex_EqHw_GetLeftGainsTarget(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b02c + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -static void vortex_EqHw_GetRightGainsTarget(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b20c + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -static void vortex_EqHw_GetLeftGainsCurrent(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b028 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -static void vortex_EqHw_GetRightGainsCurrent(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx = 0; - - if (eqhw->this04 < 0) - return; - - do { - a[ebx] = hwread(vortex->mmio, 0x2b208 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); -} - -#endif /* EQ band levels settings */ static void vortex_EqHw_SetLevels(vortex_t * vortex, u16 peaks[]) { @@ -344,38 +238,6 @@ static void vortex_EqHw_SetLevels(vortex hwwrite(vortex->mmio, 0x2b3f0, peaks[3 + (eqhw->this04 * 2)]); } -#if 0 -static void vortex_EqHw_GetLevels(vortex_t * vortex, u16 a[]) -{ - eqhw_t *eqhw = &(vortex->eq.this04); - int ebx; - - if (eqhw->this04 < 0) - return; - - ebx = 0; - do { - a[ebx] = hwread(vortex->mmio, 0x2b024 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); - - a[eqhw->this04] = hwread(vortex->mmio, 0x2b3cc); - a[eqhw->this04 + 1] = hwread(vortex->mmio, 0x2b3d8); - - ebx = 0; - do { - a[ebx + (eqhw->this04 + 2)] = - hwread(vortex->mmio, 0x2b204 + ebx * 0x30); - ebx++; - } - while (ebx < eqhw->this04); - - a[2 + (eqhw->this04 * 2)] = hwread(vortex->mmio, 0x2b3e4); - a[3 + (eqhw->this04 * 2)] = hwread(vortex->mmio, 0x2b3f0); -} - -#endif /* Global Control */ static void vortex_EqHw_SetControlReg(vortex_t * vortex, u32 reg) { @@ -387,18 +249,6 @@ static void vortex_EqHw_SetSampleRate(vo hwwrite(vortex->mmio, 0x2b440, ((sr & 0x1f) << 3) | 0xb800); } -#if 0 -static void vortex_EqHw_GetControlReg(vortex_t * vortex, u32 *reg) -{ - *reg = hwread(vortex->mmio, 0x2b440); -} - -static void vortex_EqHw_GetSampleRate(vortex_t * vortex, u32 *sr) -{ - *sr = (hwread(vortex->mmio, 0x2b440) >> 3) & 0x1f; -} - -#endif static void vortex_EqHw_Enable(vortex_t * vortex) { hwwrite(vortex->mmio, VORTEX_EQ_CTRL, 0xf001); @@ -552,29 +402,6 @@ static void vortex_Eqlzr_SetRightGain(vo vortex_EqHw_SetRightGainsSingleTarget(vortex, index, gain); } -#if 0 -static int -vortex_Eqlzr_GetAllBands(vortex_t * vortex, u16 * gains, s32 *cnt) -{ - eqlzr_t *eq = &(vortex->eq); - int si = 0; - - if (eq->this10 == 0) - return 1; - - { - if (vortex_Eqlzr_GetLeftGain(vortex, si, &gains[si])) - return 1; - if (vortex_Eqlzr_GetRightGain - (vortex, si, &gains[si + eq->this10])) - return 1; - si++; - } - while (eq->this10 > si) ; - *cnt = si * 2; - return 0; -} -#endif static int vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex_t * vortex) { eqlzr_t *eq = &(vortex->eq); @@ -678,14 +505,6 @@ static int vortex_Eqlzr_GetAllPeaks(vort return 0; } -#if 0 -static auxxEqCoeffSet_t *vortex_Eqlzr_GetActiveCoefSet(vortex_t * vortex) -{ - eqlzr_t *eq = &(vortex->eq); - - return (&(eq->coefset)); -} -#endif static void vortex_Eqlzr_init(vortex_t * vortex) { eqlzr_t *eq = &(vortex->eq);

1 0

[alsa-devel] [PATCH -mm] sound/pci/au88x0/au88x0.c remove unused code
by Michal Piotrowski 25 Mar '07

25 Mar '07

This patch is against the latest -mm snapshot. Regards, Michal -- Michal K. K. Piotrowski LTG - Linux Testers Group (PL) (http://www.stardust.webpages.pl/ltg/) LTG - Linux Testers Group (EN) (http://www.stardust.webpages.pl/linux_testers_group_en/) Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-work5/sound/pci/au88x0/au88x0.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-work3/sound/pci/au88x0/au88x0.c 2007-03-25 17:10:42.000000000 +0200 @@ -313,23 +313,6 @@ snd_vortex_probe(struct pci_dev *pci, co vortex_gameport_register(chip); -#if 0 - if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_VORTEX_SYNTH, - sizeof(snd_vortex_synth_arg_t), &wave) < 0 - || wave == NULL) { - snd_printk(KERN_ERR "Can't initialize Aureal wavetable synth\n"); - } else { - snd_vortex_synth_arg_t *arg; - - arg = SNDRV_SEQ_DEVICE_ARGPTR(wave); - strcpy(wave->name, "Aureal Synth"); - arg->hwptr = vortex; - arg->index = 1; - arg->seq_ports = seq_ports[dev]; - arg->max_voices = max_synth_voices[dev]; - } -#endif - // (5) if ((err = pci_read_config_word(pci, PCI_DEVICE_ID, &(chip->device))) < 0) {

1 0

[alsa-devel] ALSA initialization still seems to be too late?
by Adrian Bunk 23 Mar '07

23 Mar '07

The commit below from 2005 (sic) seems to be an example that workarounds often have quite a long lifetime. Can we get this sorted out properly for 2.6.22? TIA Adrian commit c7ac6b42ffba28c350cbcd48268f46689f6eb1cc Author: Linus Torvalds <torvalds(a)g5.osdl.org> Date: Wed Dec 21 14:52:32 2005 -0800 Initialize drivers/media/video/saa7134 late When compiled-in, make sure the sound system has initialized before these drivers do. Reported by Adrian Bunk <bunk(a)stusta.de> (The right fix would be to make the sound core use "subsys_initcall()" and thus initialize before all normal drivers, but this is the quick and limited safe fix for 2.6.15). Signed-off-by: Linus Torvalds <torvalds(a)osdl.org> diff --git a/drivers/media/video/saa7134/saa7134-alsa.c b/drivers/media/video/saa7134/saa7134-alsa.c index 953d5fe..6752dd1 100644 --- a/drivers/media/video/saa7134/saa7134-alsa.c +++ b/drivers/media/video/saa7134/saa7134-alsa.c @@ -1028,7 +1028,8 @@ static void saa7134_alsa_exit(void) return; } -module_init(saa7134_alsa_init); +/* We initialize this late, to make sure the sound system is up and running */ +late_initcall(saa7134_alsa_init); module_exit(saa7134_alsa_exit); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Ricardo Cerqueira"); diff --git a/drivers/media/video/saa7134/saa7134-oss.c b/drivers/media/video/saa7134/saa7134-oss.c index 513a699..c450d57 100644 --- a/drivers/media/video/saa7134/saa7134-oss.c +++ b/drivers/media/video/saa7134/saa7134-oss.c @@ -1002,7 +1002,8 @@ static void saa7134_oss_exit(void) return; } -module_init(saa7134_oss_init); +/* We initialize this late, to make sure the sound system is up and running */ +late_initcall(saa7134_oss_init); module_exit(saa7134_oss_exit); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Gerd Knorr <kraxel(a)bytesex.org> [SuSE Labs]");

1 0

[alsa-devel] [5/5] 2.6.21-rc4: known regressions (v2)
by Adrian Bunk 23 Mar '07

23 Mar '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 : Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Status : unknown Subject : USB: iPod doesn't work References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged 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> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : problem is being debugged Subject : forcedeth: skb_over_panic References : http://bugzilla.kernel.org/show_bug.cgi?id=8058 Submitter : Albert Hopkins <kernel(a)marduk.letterboxes.org> Handled-By : Ayaz Abdulla <aabdulla(a)nvidia.com> Patch : http://bugzilla.kernel.org/show_bug.cgi?id=8058 Status : patch available

1 0

[alsa-devel] snd_pcm_writei() error: Write error: -32,Broken pipe
by John Utz 23 Mar '07

23 Mar '07

Hello; I am running into some trouble with writing out to the device, i get some sound and then i get the Write error. Heres the code fragment: if (-EAGAIN == (iErr = snd_pcm_writei(handle, pu8Frames, iSizPer))) continue; if (0 > iErr) { printf("Write error: %d,%s\n", iErr, snd_strerror(iErr)); Is the problem that my buffer (pointed to by pu8frames) isnt big enuf? or is this a different failure entirely. If so, what are the possibilities? tnx!

1 0

[alsa-devel] PCM -- no need to release params mem nor close handle?
by intellij 23 Mar '07

23 Mar '07

I am looking at PCM interface manual http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html It tells that there are an open and set_params functions but nothing about close(). Furthermore, the description of open() http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#g8340c7dc0… neither tells that. Ok, I have realized from the examples. However, even the examples demonstrate no params deallocation. You call the snd_pcm_hw_params_alloca(&hwparams) but never release the allocated memory. Why? Thank you.

3 2

[alsa-devel] Supported architecture: m68knommu?
by Sven Johnsson 23 Mar '07

23 Mar '07

Hello, I am interested in building ALSA for the m68knommu architecture on uClinux. Is this possible? Best Regards, Sven

2 1

Re: [alsa-devel] Réf. : Where is _snd_pcm defined?
by intellij 23 Mar '07

23 Mar '07

Thank you for your answer. However, you information is not correct. I do not have the snd_pcm structure in the three copies of pcm.h I have in alsa-lib/include, arm-linux/alsa nor arm-linux/include. Peahaps, this is because my lib is slightly obsolete. Fortunately, today I have accidentally discovered the requested _snd_pcm structure in the alsa-lib/src/pcm/pcm_local.h. Excuse me for the fuss. BTW, we have fallen out of the mailing list. Forums are efficient only when there are many participants for two reasons: neighbors can provide more information and the same questions will be prevented from asking repeatedly. Do you think this case is different? > >Hello, > >If you get the alsa-driver sources, you can have everything you need (hope :-). >Note that in my typedefs.h, I don't have any _ at the beginin, so "struct snd_pcm" is in alsa-kernel/include/pcm.h. > >Regards, >JM > > >********************************* >This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised >use or dissemination is prohibited. >Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or >falsified. >If you are not the intended addressee of this message, please cancel it immediately and inform the sender. >******************************** -- Яндекс.Почта: объем почтового ящика не ограничен! http://mail.yandex.ru/monitoring/

2 1

[alsa-devel] Where is _snd_pcm defined?
by intellij 22 Mar '07

22 Mar '07

Hello, I have gone through all the source codes to wee the definition of snd_pcm_t and see that it is defined typedef struct _snd_pcm snd_pcm_t; But is has many fields inside! At least, name and fast_ops pointers should be there. I do not understand where are they hiding. Actually, I am an alsa user. However, the system occasionaly hangs on writing into the buffer (because the consumption stops). For this reason, I'm looking for the features for hacking the system. I have an idea of turning the printfs on/of in runtime by a shared flag. I could put the pointer to the flag into pcm, so that my application could modify it and the library functions would check it before printing the traces. Thank you.

1 0

[alsa-devel] (no subject)
by ODINOT Jean-Marc ETRALI 22 Mar '07

22 Mar '07

Hi Alsa gurus, I've to write an alsa driver for a custom card. I readed "writing an alsa driver" and other paper about alsa and drivers. Well I got a result, but not as good as I think it's possible with alsa. My card have 30 output slots (and 7 input slots), can be 8Khz or 16Khz (same rate for every slots), 16bits little endian by sample, can have one or more sample for each slot (same for every slots) in a frame, an hardware IT every frame (a frame is 30 slots :-). The purpose is to get, for each IT, a frame with all the datas. IT -> frame [channel1][channel2][channel3]...[channel30] channelX [sample1][sample2]...[sampleN] sampleX [16bitsle] At this time, I've something like a frame if I use channel30 with speaker-test -c30, but if I use channel1, alsa give me a buffer with many samples for channel1. And, just after oppening, I get an extra word. BTW I use .copy to get the buffer. My questions: -why the extra word (can I avoid it?)? -how set the buffer len to always have all my channels (used or not)? -is it possible to have more than a sample by channel contiguous? Think's it's all for today, but maybe your answers will give me more ideas :-) Thanx for replies (and please, forgive my awful english :-(, JM PS: sorry for my previous (and my first) post which was in html only ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

[alsa-devel] pcm low level
by ODINOT Jean-Marc ETRALI 22 Mar '07

22 Mar '07

********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

Re: [alsa-devel] [Alsa-devel] Static build of alsa-lib
by Takashi Iwai 22 Mar '07

22 Mar '07

At Tue, 20 Mar 2007 08:13:41 -0700 (PDT), Ciaccia wrote: > > Hi Takashi, > It's clear that the concept of dynamic loading doesn't > conflict with static library, I think my original > question was not completely clear. > > I would like to port a sound application (just PCM, no > midi or other stuff) to an embedded system. The core > of this embedded system is a Cirrus EP9302 ARM > processor, which supports hardware floating points. > > ARM processors have several ABIs (EABI, OABI, ???) and > programs compiled using one ABI are not linkable > (neither at compile time, nor at run time) with other > binaries, because of the different format. This is a > problem for binary distributions (such as Debian), > since the same binary does not work on all ARM > architectures. > > For a reason I really don't understand, EP9302 > hardware floating point binaries only work when > compiled with -static (don't ask me why...), and > therefore I wanted to have a "static" (=without shared > libraries) ALSA application. If this would not be > possible I can always use OSS (which does not requires > shared objects to be loaded at run time), but I still > think it should be possible to develop ALSA > applications for architectures where dynamic loading > is not available... > > Is there a way to achieve this? Sure, what I meant is that the patch had no function to disable the libdl and libpthread explicitly but only checked. The new patch below, for example, can give you options --with-libdl and --with-pthread. For disabling libdl, pass --with-libdl=no. If this works for you, I'll apply it to the upstream. Takashi diff -r 6d0a999aef24 Makefile.am --- a/Makefile.am Fri Mar 16 15:22:27 2007 +0100 +++ b/Makefile.am Tue Mar 20 16:55:23 2007 +0100 @@ -1,4 +1,7 @@ SUBDIRS=doc include src modules -SUBDIRS=doc include src modules +SUBDIRS=doc include src +if BUILD_MODULES +SUBDIRS += modules +endif if BUILD_PCM_PLUGIN_SHM SUBDIRS += aserver endif diff -r 6d0a999aef24 configure.in --- a/configure.in Fri Mar 16 15:22:27 2007 +0100 +++ b/configure.in Tue Mar 20 16:55:00 2007 +0100 @@ -148,6 +148,44 @@ else else AC_MSG_RESULT(no) fi + +ALSA_DEPLIBS="" +if test "$softfloat" != "yes"; then + ALSA_DEPLIBS="-lm" +fi + +dnl Check for libdl +AC_MSG_CHECKING(for libdl) +AC_ARG_WITH(libdl, + [ --with-libdl Use libdl for plugins (default = yes)], + [ have_libdl="$withval" ], [ have_libdl="yes" ]) +if test "$have_libdl" = "yes"; then + AC_CHECK_LIB([dl], [dlsym], [HAVE_LIBDL="yes"]) + if test "$HAVE_LIBDL" = "yes" ; then + ALSA_DEPLIBS="$ALSA_DEPLIBS -ldl" + AC_DEFINE([HAVE_LIBDL], 1, [Have libdl]) + fi +else + AC_MSG_RESULT(no) +fi +AM_CONDITIONAL(BUILD_MODULES, test "$HAVE_LIBDL"="yes") + +dnl Check for pthread +AC_MSG_CHECKING(for pthread) +AC_ARG_WITH(pthread, + [ --with-pthread Use pthread (default = yes)], + [ have_pthread="$withval" ], [ have_pthread="yes" ]) +if test "$have_pthread" = "yes"; then + AC_CHECK_LIB([pthread], [pthread_join], [HAVE_LIBPTHREAD="yes"]) + if test "$HAVE_LIBPTHREAD" = "yes"; then + ALSA_DEPLIBS="$ALSA_DEPLIBS -lpthread" + AC_DEFINE([HAVE_LIBPTHREAD], 1, [Have libpthread]) + fi +else + AC_MSG_RESULT(no) +fi + +AC_SUBST(ALSA_DEPLIBS) dnl Check for architecture AC_MSG_CHECKING(for architecture) @@ -318,6 +356,21 @@ fi if test "$build_pcm_ioplug" = "yes"; then build_pcm_extplug="yes" +fi + +if test "$HAVE_LIBDL" != "yes"; then + build_pcm_meter="no" + build_pcm_ladspa="no" + build_pcm_pcm_ioplug="no" + build_pcm_pcm_extplug="no" +fi + +if test "$HAVE_LIBPTHREAD" != "yes"; then + build_pcm_share="no" +fi + +if test "$softfloat" != "yes"; then + build_pcm_lfloat="no" fi AM_CONDITIONAL(BUILD_PCM_PLUGIN, test x$build_pcm_plugin = xyes) diff -r 6d0a999aef24 include/local.h --- a/include/local.h Fri Mar 16 15:22:27 2007 +0100 +++ b/include/local.h Tue Mar 20 16:16:36 2007 +0100 @@ -36,6 +36,11 @@ #include "config.h" #ifdef SUPPORT_RESMGR #include <resmgr.h> +#endif +#ifdef HAVE_LIBDL +#include <dlfcn.h> +#else +#define RTLD_NOW 0 #endif #define _snd_config_iterator list_head diff -r 6d0a999aef24 src/Makefile.am --- a/src/Makefile.am Fri Mar 16 15:22:27 2007 +0100 +++ b/src/Makefile.am Tue Mar 20 16:16:36 2007 +0100 @@ -41,7 +41,7 @@ libasound_la_LIBADD += alisp/libalisp.la libasound_la_LIBADD += alisp/libalisp.la endif SUBDIRS += compat conf -libasound_la_LIBADD += compat/libcompat.la -lm -ldl -lpthread +libasound_la_LIBADD += compat/libcompat.la @ALSA_DEPLIBS@ libasound_la_LDFLAGS = -version-info $(COMPATNUM) $(VSYMS) diff -r 6d0a999aef24 src/async.c --- a/src/async.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/async.c Tue Mar 20 16:16:36 2007 +0100 @@ -151,9 +151,11 @@ int snd_async_del_handler(snd_async_hand if (!list_empty(&handler->hlist)) goto _end; switch (handler->type) { +#ifdef BUILD_PCM case SND_ASYNC_HANDLER_PCM: err = snd_pcm_async(handler->u.pcm, -1, 1); break; +#endif case SND_ASYNC_HANDLER_CTL: err = snd_ctl_async(handler->u.ctl, -1, 1); break; diff -r 6d0a999aef24 src/conf.c --- a/src/conf.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/conf.c Tue Mar 20 16:16:36 2007 +0100 @@ -415,12 +415,13 @@ beginning:</P> #include <stdarg.h> -#include <dlfcn.h> #include <limits.h> #include <sys/stat.h> -#include <pthread.h> #include <locale.h> #include "local.h" +#ifdef HAVE_LIBPTHREAD +#include <pthread.h> +#endif #ifndef DOC_HIDDEN @@ -3080,7 +3081,9 @@ int snd_config_update_r(snd_config_t **_ return 1; } +#ifdef HAVE_LIBPTHREAD static pthread_mutex_t snd_config_update_mutex = PTHREAD_MUTEX_INITIALIZER; +#endif /** * \brief Updates #snd_config by rereading the global configuration files (if needed). @@ -3099,9 +3102,13 @@ int snd_config_update(void) { int err; +#ifdef HAVE_LIBPTHREAD pthread_mutex_lock(&snd_config_update_mutex); +#endif err = snd_config_update_r(&snd_config, &snd_config_global_update, NULL); +#ifdef HAVE_LIBPTHREAD pthread_mutex_unlock(&snd_config_update_mutex); +#endif return err; } @@ -3128,15 +3135,18 @@ int snd_config_update_free(snd_config_up */ int snd_config_update_free_global(void) { +#ifdef HAVE_LIBPTHREAD pthread_mutex_lock(&snd_config_update_mutex); +#endif if (snd_config) snd_config_delete(snd_config); snd_config = NULL; if (snd_config_global_update) snd_config_update_free(snd_config_global_update); snd_config_global_update = NULL; +#ifdef HAVE_LIBPTHREAD pthread_mutex_unlock(&snd_config_update_mutex); - +#endif /* FIXME: better to place this in another place... */ snd_dlobj_cache_cleanup(); diff -r 6d0a999aef24 src/confmisc.c --- a/src/confmisc.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/confmisc.c Tue Mar 20 16:16:36 2007 +0100 @@ -946,6 +946,8 @@ SND_DLSYM_BUILD_VERSION(snd_func_card_na SND_DLSYM_BUILD_VERSION(snd_func_card_name, SND_CONFIG_DLSYM_VERSION_EVALUATE); #endif +#ifdef BUILD_PCM + /** * \brief Returns the pcm identification of a device. * \param dst The function puts the handle to the result configuration node @@ -1198,6 +1200,8 @@ int snd_func_private_pcm_subdevice(snd_c #ifndef DOC_HIDDEN SND_DLSYM_BUILD_VERSION(snd_func_private_pcm_subdevice, SND_CONFIG_DLSYM_VERSION_EVALUATE); #endif + +#endif /* BUILD_PCM */ /** * \brief Copies the specified configuration node. diff -r 6d0a999aef24 src/control/control.c --- a/src/control/control.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/control/control.c Tue Mar 20 16:16:36 2007 +0100 @@ -47,7 +47,6 @@ and IEC958 structure. #include <string.h> #include <fcntl.h> #include <signal.h> -#include <dlfcn.h> #include <sys/poll.h> #include "control_local.h" diff -r 6d0a999aef24 src/control/hcontrol.c --- a/src/control/hcontrol.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/control/hcontrol.c Tue Mar 20 16:16:36 2007 +0100 @@ -48,11 +48,13 @@ to reduce overhead accessing the real co #include <string.h> #include <fcntl.h> #include <sys/ioctl.h> -#include <pthread.h> #ifndef DOC_HIDDEN #define __USE_GNU #endif #include "control_local.h" +#ifdef HAVE_LIBPTHREAD +#include <pthread.h> +#endif #ifndef DOC_HIDDEN #define NOT_FOUND 1000000000 @@ -420,17 +422,22 @@ static void snd_hctl_sort(snd_hctl_t *hc static void snd_hctl_sort(snd_hctl_t *hctl) { unsigned int k; +#ifdef HAVE_LIBPTHREAD static pthread_mutex_t sync_lock = PTHREAD_MUTEX_INITIALIZER; +#endif assert(hctl); assert(hctl->compare); INIT_LIST_HEAD(&hctl->elems); +#ifdef HAVE_LIBPTHREAD pthread_mutex_lock(&sync_lock); +#endif compare_hctl = hctl; qsort(hctl->pelems, hctl->count, sizeof(*hctl->pelems), hctl_compare); +#ifdef HAVE_LIBPTHREAD pthread_mutex_unlock(&sync_lock); - +#endif for (k = 0; k < hctl->count; k++) list_add_tail(&hctl->pelems[k]->list, &hctl->elems); } diff -r 6d0a999aef24 src/dlmisc.c --- a/src/dlmisc.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/dlmisc.c Tue Mar 20 16:51:13 2007 +0100 @@ -28,7 +28,6 @@ */ #define _GNU_SOURCE -#include <dlfcn.h> #include "list.h" #include "local.h" @@ -53,13 +52,19 @@ void *snd_dlopen(const char *name, int m if (name == NULL) return &snd_dlsym_start; #else +#ifdef HAVE_LIBDL if (name == NULL) { Dl_info dlinfo; if (dladdr(snd_dlopen, &dlinfo) > 0) name = dlinfo.dli_fname; } #endif +#endif +#ifdef HAVE_LIBDL return dlopen(name, mode); +#else + return NULL; +#endif } /** @@ -76,7 +81,11 @@ int snd_dlclose(void *handle) if (handle == &snd_dlsym_start) return 0; #endif +#ifdef HAVE_LIBDL return dlclose(handle); +#else + return 0; +#endif } /** @@ -91,6 +100,7 @@ int snd_dlclose(void *handle) */ static int snd_dlsym_verify(void *handle, const char *name, const char *version) { +#ifdef HAVE_LIBDL int res; char *vname; @@ -107,6 +117,9 @@ static int snd_dlsym_verify(void *handle if (res < 0) SNDERR("unable to verify version for symbol %s", name); return res; +#else + return 0; +#endif } /** @@ -139,10 +152,16 @@ void *snd_dlsym(void *handle, const char return NULL; } #endif - err = snd_dlsym_verify(handle, name, version); - if (err < 0) - return NULL; +#ifdef HAVE_LIBDL + if (version) { + err = snd_dlsym_verify(handle, name, version); + if (err < 0) + return NULL; + } return dlsym(handle, name); +#else + return NULL; +#endif } /* diff -r 6d0a999aef24 src/hwdep/hwdep.c --- a/src/hwdep/hwdep.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/hwdep/hwdep.c Tue Mar 20 16:16:36 2007 +0100 @@ -33,7 +33,6 @@ #include <unistd.h> #include <string.h> #include <fcntl.h> -#include <dlfcn.h> #include <sys/ioctl.h> #include "hwdep_local.h" diff -r 6d0a999aef24 src/mixer/simple_abst.c --- a/src/mixer/simple_abst.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/mixer/simple_abst.c Tue Mar 20 16:52:35 2007 +0100 @@ -82,14 +82,14 @@ static int try_open(snd_mixer_class_t *c free(xlib); return -ENXIO; } - event_func = dlsym(h, "alsa_mixer_simple_event"); + event_func = snd_dlsym(h, "alsa_mixer_simple_event", NULL); if (event_func == NULL) { SNDERR("Symbol 'alsa_mixer_simple_event' was not found in '%s'", xlib); snd_dlclose(h); free(xlib); return -ENXIO; } - init_func = dlsym(h, "alsa_mixer_simple_init"); + init_func = snd_dlsym(h, "alsa_mixer_simple_init", NULL); if (init_func == NULL) { SNDERR("Symbol 'alsa_mixer_simple_init' was not found in '%s'", xlib); snd_dlclose(h); diff -r 6d0a999aef24 src/pcm/pcm.c --- a/src/pcm/pcm.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm.c Tue Mar 20 16:16:36 2007 +0100 @@ -634,7 +634,6 @@ playback devices. #include <malloc.h> #include <stdarg.h> #include <signal.h> -#include <dlfcn.h> #include <sys/poll.h> #include <sys/shm.h> #include <sys/mman.h> diff -r 6d0a999aef24 src/pcm/pcm_hooks.c --- a/src/pcm/pcm_hooks.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm_hooks.c Tue Mar 20 16:16:36 2007 +0100 @@ -27,7 +27,6 @@ * */ -#include <dlfcn.h> #include "pcm_local.h" #include "pcm_generic.h" diff -r 6d0a999aef24 src/pcm/pcm_ladspa.c --- a/src/pcm/pcm_ladspa.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm_ladspa.c Tue Mar 20 16:16:36 2007 +0100 @@ -33,7 +33,6 @@ */ #include <dirent.h> -#include <dlfcn.h> #include <locale.h> #include <math.h> #include "pcm_local.h" diff -r 6d0a999aef24 src/pcm/pcm_rate.c --- a/src/pcm/pcm_rate.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/pcm/pcm_rate.c Tue Mar 20 16:51:41 2007 +0100 @@ -29,7 +29,6 @@ */ #include <inttypes.h> #include <byteswap.h> -#include <dlfcn.h> #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" @@ -1326,7 +1325,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, free(rate); return -ENOENT; } - open_func = dlsym(h, open_name); + open_func = snd_dlsym(h, open_name, NULL); if (! open_func) { SNDERR("Cannot find function %s", open_name); snd_dlclose(h); diff -r 6d0a999aef24 src/rawmidi/rawmidi.c --- a/src/rawmidi/rawmidi.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/rawmidi/rawmidi.c Tue Mar 20 16:16:36 2007 +0100 @@ -139,7 +139,6 @@ This example shows open and read/write r #include <stdarg.h> #include <unistd.h> #include <string.h> -#include <dlfcn.h> #include "rawmidi_local.h" /** diff -r 6d0a999aef24 src/seq/seq.c --- a/src/seq/seq.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/seq/seq.c Tue Mar 20 16:16:36 2007 +0100 @@ -777,7 +777,6 @@ void event_filter(snd_seq_t *seq, snd_se */ -#include <dlfcn.h> #include <sys/poll.h> #include "seq_local.h" diff -r 6d0a999aef24 src/timer/timer.c --- a/src/timer/timer.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/timer/timer.c Tue Mar 20 16:16:36 2007 +0100 @@ -72,7 +72,6 @@ This example shows opening a timer devic #include <unistd.h> #include <string.h> #include <fcntl.h> -#include <dlfcn.h> #include <signal.h> #include <sys/ioctl.h> #include "timer_local.h" diff -r 6d0a999aef24 src/timer/timer_query.c --- a/src/timer/timer_query.c Fri Mar 16 15:22:27 2007 +0100 +++ b/src/timer/timer_query.c Tue Mar 20 16:16:36 2007 +0100 @@ -31,7 +31,6 @@ #include <unistd.h> #include <string.h> #include <fcntl.h> -#include <dlfcn.h> #include <sys/ioctl.h> #include "timer_local.h" diff -r 6d0a999aef24 utils/alsa.pc.in --- a/utils/alsa.pc.in Fri Mar 16 15:22:27 2007 +0100 +++ b/utils/alsa.pc.in Tue Mar 20 16:16:36 2007 +0100 @@ -8,7 +8,7 @@ Version: @VERSION@ Version: @VERSION@ Requires: Libs: -L${libdir} -lasound -Libs.private: -lm -ldl -lpthread +Libs.private: @ALSA_DEPLIBS@ # -I${includedir}/alsa below is just for backward compatibility # (it was set so mistakely in the older version) Cflags: -I${includedir} -I${includedir}/alsa

2 3

[alsa-devel] During ADPCM playback, when is the reference byte set?
by Rask Ingemann Lambertsen 21 Mar '07

21 Mar '07

Hi. When an app is playing ADPCM samples, how does the driver know how to find the reference byte, other than assuming it is the first byte of sample data? Could there be more than one? Etc. -- Rask Ingemann Lambertsen

1 0

[alsa-devel] s/sf.net/alsa-devel.org/
by Rene Herman 21 Mar '07

21 Mar '07

Hi Takashi. This replaces all occurences of alsa-devel(a)lists.s[ource]f[orge].net that a simple recursive grep found in the current HG ALSA repos by alsa-devel(a)alsa-project.org. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff -urN alsa.orig/alsa-driver/pci/bt87x.patch alsa/alsa-driver/pci/bt87x.patch --- alsa.orig/alsa-driver/pci/bt87x.patch 2006-06-29 23:22:39.000000000 +0200 +++ alsa/alsa-driver/pci/bt87x.patch 2007-03-20 17:52:59.000000000 +0100 @@ -39,7 +39,7 @@ + pci->device, subsystem_vendor, subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " - "<alsa-devel(a)lists.sf.net>\n"); + "<alsa-devel(a)alsa-project.org>\n"); @@ -935,3 +948,5 @@ module_init(alsa_card_bt87x_init) diff -urN alsa.orig/alsa-kernel/Documentation/ALSA-Configuration.txt alsa/alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa.orig/alsa-kernel/Documentation/ALSA-Configuration.txt 2007-02-26 17:00:05.000000000 +0100 +++ alsa/alsa-kernel/Documentation/ALSA-Configuration.txt 2007-03-20 17:54:24.000000000 +0100 @@ -2049,4 +2049,4 @@ https://bugtrack.alsa-project.org/bugs/ ALSA Developers ML - mailto:alsa-devel@lists.sourceforge.net + mailto:alsa-devel@alsa-project.org diff -urN alsa.orig/alsa-kernel/Documentation/Bt87x.txt alsa/alsa-kernel/Documentation/Bt87x.txt --- alsa.orig/alsa-kernel/Documentation/Bt87x.txt 2006-06-29 23:25:22.000000000 +0200 +++ alsa/alsa-kernel/Documentation/Bt87x.txt 2007-03-20 17:55:06.000000000 +0100 @@ -36,8 +36,8 @@ other values than the default 32000 (often it's 44100 or 64000). If you have an unknown card, please mail the ID and board name to -<alsa-devel(a)lists.sf.net>, regardless of whether audio capture works or -not, so that future versions of this driver know about your card. +<alsa-devel(a)alsa-project.org>, regardless of whether audio capture works +or not, so that future versions of this driver know about your card. Audio modes diff -urN alsa.orig/alsa-kernel/pci/bt87x.c alsa/alsa-kernel/pci/bt87x.c --- alsa.orig/alsa-kernel/pci/bt87x.c 2007-03-06 11:30:32.000000000 +0100 +++ alsa/alsa-kernel/pci/bt87x.c 2007-03-20 17:56:40.000000000 +0100 @@ -835,7 +835,7 @@ pci->device, pci->subsystem_vendor, pci->subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " - "<alsa-devel(a)lists.sf.net>\n"); + "<alsa-devel(a)alsa-project.org>\n"); return 32000; /* default rate */ } diff -urN alsa.orig/alsa-kernel/pci/cs46xx/cs46xx_lib.c alsa/alsa-kernel/pci/cs46xx/cs46xx_lib.c --- alsa.orig/alsa-kernel/pci/cs46xx/cs46xx_lib.c 2006-11-21 18:00:06.000000000 +0100 +++ alsa/alsa-kernel/pci/cs46xx/cs46xx_lib.c 2007-03-20 17:55:54.000000000 +0100 @@ -3107,7 +3107,7 @@ snd_printk(KERN_ERR "ERROR: snd-cs46xx: never read ISV3 & ISV4 from AC'97\n"); snd_printk(KERN_ERR " Try reloading the ALSA driver, if you find something\n"); snd_printk(KERN_ERR " broken or not working on your soundcard upon\n"); - snd_printk(KERN_ERR " this message please report to alsa-devel(a)lists.sourceforge.net\n"); + snd_printk(KERN_ERR " this message please report to alsa-devel(a)alsa-project.org\n"); return -EIO; #endif diff -urN alsa.orig/alsa-lib/src/conf/cards/USB-Audio.conf alsa/alsa-lib/src/conf/cards/USB-Audio.conf --- alsa.orig/alsa-lib/src/conf/cards/USB-Audio.conf 2007-02-13 10:00:05.000000000 +0100 +++ alsa/alsa-lib/src/conf/cards/USB-Audio.conf 2007-03-20 17:57:20.000000000 +0100 @@ -9,7 +9,7 @@ # cards.USB-Audio.pcm.use_dmix."NoiseBlaster 3000" no # # If your device requires such a definition to work correctly, please report it -# to <alsa-devel(a)lists.sourceforge.net>. +# to <alsa-devel(a)alsa-project.org>. # If a device has sample formats not supported by dmix, dmix can be disabled

2 1

[alsa-devel] Problem in dummy driver
by Himanshu Chauhan 21 Mar '07

21 Mar '07

Dear all, I am working on the dummy driver provided with the ALSA. I took it from the linux kernel 2.6.20.1 I build the module and load it. The XMMS seems to play ok (doesn't hang and all) but none of the recording application seem to record from the driver. Can the driver in current state work as the loop back cable between applications? Thanks in advance... Warm Regards --Himanshu

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[alsa-devel] RESUBSCRIBE - New alsa-devel mailing list
by Jaroslav Kysela 21 Mar '07

21 Mar '07

Hello everyone, because never ending story with SF services, I quickly setup new mailman alsa-devel list on new ALSA server (we'll get working list archive as bonus). Because I don't want to move automatically all subscribers, please, subscribe yourself to new list: http://mailman.alsa-project.org/mailman/listinfo/alsa-devel New list is available as alsa-devel(a)alsa-project.org while old on SF is available as alsa-devel(a)lists.sourceforge.net . The archive for new alsa-devel mailing list is here: http://mailman.alsa-project.org/pipermail/alsa-devel/ The posts from non-members are moderated now! Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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Re: [alsa-devel] Development Question
by M. Gagnon 21 Mar '07

21 Mar '07

> > >What is the output of the program? > > > > pending events size: 28 > > pending events size: 56 > > pending events size: 84 > > ... > > BEFORE DRAIN: pending events size: 476 > > AFTER DRAIN: pending events size: 0 > >Looks OK. In theory, all events should be sent somewhere. > >Please run aseqdump and send the events to its port. What does it >output? > % aseqdump Waiting for data at port 128:0. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 0:1 Port subscribed 129:0 -> 128:0 129:0 Note on 2 60 127 0:1 Port unsubscribed 129:0 -> 128:0 Seems like only one event gets sent - just like what i hear. >Run "cat /proc/asound/seq/queues" and >"cat /proc/asound/seq/timer". > % cat /proc/asound/seq/queues queue 0: [Queue-0] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 96 current tempo : 500000 current BPM : 120 current time : 1.536000000 s current tick : 294 queue 1: [Queue-1] owned by client : 129 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 48 current tempo : 1000000 current BPM : 60 current time : 1.488000000 s current tick : 71 % cat /proc/asound/seq/timer Timer for queue 0 : system timer Period time : 0.004000000 Skew : 65536 / 65536 Timer for queue 1 : system timer Period time : 0.004000000 Skew : 65536 / 65536 thanks for your help just as reminder (so the output can be interpreted) this is my latest code /* all: g++ midi_receiver.cpp -o AlsaNoteTest -lasound */ #include <stdio.h> #include <alsa/asoundlib.h> #include "midi_receiver.h" #include <vector> #include <iostream> #include <unistd.h> #include <pthread.h> pthread_t thread1; char *message1 = "Thread 1"; int iret1; void *play_func( void *ptr ) { AlsaNotePlayer::play(); } int main() { AlsaNotePlayer::findDevices(); AlsaNotePlayer::openDevice(128, 0); AlsaNotePlayer::setTempo(60, 48); int time=0; int duration = 50; AlsaNotePlayer::noteOn_seq(1, 60, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 60, 0, time); AlsaNotePlayer::noteOn_seq(1, 62, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 62, 0, time); AlsaNotePlayer::noteOn_seq(1, 60, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 60, 0, time); AlsaNotePlayer::noteOn_seq(1, 59, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 59, 0, time); // ---------- AlsaNotePlayer::noteOn_seq(1, 63, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 63, 0, time); AlsaNotePlayer::noteOn_seq(1, 65, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 65, 0, time); AlsaNotePlayer::noteOn_seq(1, 63, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 63, 0, time); AlsaNotePlayer::noteOn_seq(1, 61, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 61, 0, time); iret1 = pthread_create( &thread1, NULL, play_func, (void*) message1); pthread_join( thread1, NULL); while(1) { usleep(10000); } } namespace AlsaNotePlayer { class MidiDevice { public: int client, port; char* name; MidiDevice(int client_arg, int port_arg, const char* name_arg) { client = client_arg; port = port_arg; name = (char*)name_arg; } }; std::vector<MidiDevice> devices; struct MidiContext { snd_seq_t* sequencer; int queue; snd_seq_addr_t address; int port; snd_seq_port_subscribe_t *subs; }; struct MidiContext MidiContext; snd_seq_addr_t address; void findDevices() { snd_seq_client_info_t* clientInfo; if (snd_seq_open(&MidiContext.sequencer, "default", SND_SEQ_OPEN_OUTPUT, 0) < 0) { return; } snd_seq_client_info_alloca(&clientInfo); snd_seq_client_info_set_client(clientInfo, -1); // iterate through clients while (snd_seq_query_next_client(MidiContext.sequencer, clientInfo) >= 0) { snd_seq_port_info_t* pinfo; snd_seq_port_info_alloca(&pinfo); snd_seq_port_info_set_client(pinfo, snd_seq_client_info_get_client(clientInfo)); snd_seq_port_info_set_port(pinfo, -1); // and now through ports while (snd_seq_query_next_port(MidiContext.sequencer, pinfo) >= 0) { unsigned int capability = snd_seq_port_info_get_capability(pinfo); if ((capability & SND_SEQ_PORT_CAP_SUBS_WRITE) == 0) { continue; } int client = (snd_seq_port_info_get_addr(pinfo))->client; int port = (snd_seq_port_info_get_addr(pinfo))->port; std::cout << client << ":" << port << " --> " << snd_seq_port_info_get_name(pinfo) << std::endl; devices.push_back( MidiDevice(client, port, snd_seq_port_info_get_name(pinfo)) ); } } if (snd_seq_set_client_name(MidiContext.sequencer, "Aria") < 0) { return; } //MidiContext.address.port = snd_seq_create_simple_port(MidiContext.sequencer, "Aria Port 0", SND_SEQ_PORT_CAP_SUBS_WRITE, SND_SEQ_PORT_TYPE_APPLICATION); MidiContext.address.port = snd_seq_create_simple_port(MidiContext.sequencer, "Aria Port 0", SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_READ, SND_SEQ_PORT_TYPE_APPLICATION); //SND_SEQ_PORT_TYPE_MIDI_GENERIC); MidiContext.address.client = snd_seq_client_id (MidiContext.sequencer); MidiContext.queue = snd_seq_alloc_queue (MidiContext.sequencer); snd_seq_set_client_pool_output (MidiContext.sequencer, 1024); } void setTempo(int tempo_arg, int resolution) { // argument is not yet considered snd_seq_queue_tempo_t *tempo; snd_seq_queue_tempo_alloca(&tempo); // snd_seq_queue_tempo_set_tempo(tempo, 1000000); // 60 BPM // snd_seq_queue_tempo_set_ppq(tempo, 48); // 48 PPQ snd_seq_queue_tempo_set_ppq(tempo, resolution); snd_seq_queue_tempo_set_tempo(tempo, 60*1000000/tempo_arg); snd_seq_set_queue_tempo(MidiContext.sequencer, MidiContext.queue, tempo); } void openDevice(int client, int port) { address.client = client; address.port = port; snd_seq_port_subscribe_alloca(&MidiContext.subs); snd_seq_port_subscribe_set_sender(MidiContext.subs, &MidiContext.address); snd_seq_port_subscribe_set_dest(MidiContext.subs, &address); snd_seq_subscribe_port(MidiContext.sequencer, MidiContext.subs); } void closeDevice() { snd_seq_port_subscribe_alloca(&MidiContext.subs); snd_seq_port_subscribe_set_sender(MidiContext.subs, &MidiContext.address); snd_seq_port_subscribe_set_dest(MidiContext.subs, &address); if (snd_seq_unsubscribe_port(MidiContext.sequencer, MidiContext.subs) < 0) { return; } //snd_seq_port_subscribe_free(MidiContext.subs); } //************************************************************************************// //************************************MIDI EVENTS*************************************// //************************************************************************************// void noteOn_seq(int channel, int note, int velocity, int tick) { snd_seq_event_t event; snd_seq_ev_clear(&event); event.queue = MidiContext.queue; event.source = MidiContext.address; snd_seq_ev_set_subs(&event); snd_seq_ev_set_source(&event, MidiContext.port); snd_seq_ev_set_noteon(&event, channel, note, velocity); snd_seq_ev_schedule_tick(&event, MidiContext.queue, 0, tick ); snd_seq_event_output(MidiContext.sequencer, &event); std::cout << "pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; } void noteOff_seq(int channel, int note, int velocity, int tick) { snd_seq_event_t event; snd_seq_ev_clear(&event); event.queue = MidiContext.queue; event.source = MidiContext.address; snd_seq_ev_set_subs(&event); snd_seq_ev_set_noteoff(&event, channel, note, velocity); snd_seq_ev_schedule_tick(&event, MidiContext.queue, 0 /*absolute*/, tick ); snd_seq_event_output(MidiContext.sequencer, &event); std::cout << "pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; } void play() { snd_seq_start_queue(MidiContext.sequencer, MidiContext.queue, 0); std::cout << "BEFORE DRAIN: pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; int output = snd_seq_drain_output(MidiContext.sequencer); std::cout << "AFTER DRAIN: pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; if(output == 0) std::cout << "all events drained" << std::endl; if(output > 0) std::cout << output << "bytes remaining" << std::endl; if(output < 0) { fprintf (stderr, "Error: (%s)\n", snd_strerror (output)); //std::cout << "error" << std::endl; } } } _________________________________________________________________ Ne perdez pas de temps dans les files dattente magasinez en ligne. http://magasiner.sympatico.msn.ca

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Re: [alsa-devel] [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards
by Rene Herman 20 Mar '07

20 Mar '07

On 03/19/2007 11:46 PM, Rask Ingemann Lambertsen wrote: >> This setup also shares that same problem with the BIOS generally >> not having reserved/routed the resources -- on a general PCI/ISA >> system the user needs to go into the BIOS setup and reserve the >> resources the card is (or will be, in this case) set to. So now the >> user really needs to know what resources are going to be assigned, >> destroying the plug and play idea... > > Not really. A standards compliant PnPBIOS set up with "PnP aware OS = > YES" will use only enough resources to boot the system, which should > leave enough irqs and dmas for the Linux PnP layer to do its work. There's also the "routed" bit though. I actually ran into this recently when I was playing with snd-sgalaxy (which is why I'm harping on it). An sgalaxy is also a (partly) software configurable card and when not given explicit parameters the driver just searches for an unrequested IRQ and DMA and then programs the card for it. Over here it grabbed IRQ 10 but then didn't work anyway, and only started working when I set IRQ 10 to "Legacy ISA" in the BIOS setup. It seems the IRQ simply was not available to ISA without having done that (no, no PCI cards were using IRQ 10 or anything). I must say though that my comment above was partly based on a somewhat foggy understanding of how things get assigned. Yes, _if_ this "routing" problem I seemed to be experiencing is not a serious problem in practice (Adam? Are you able to comment?) then the fact that the PnP layer will be handing out the resources taking into account the other (known) cards is indeed a practical advantage of using that layer. > I think there are three good reasons to use the PnP layer: > > 1) The PnP layer informs the PCI layer of irqs to try to avoid - please > see pnp_register_irq_resource() in drivers/pnp/resource.c. How does that help? On my systems (one AMI, all others Award BIOses) all PCI cards get assigned their IRQ before the OS ever boots. That "PnP OS: Yes/No" switch only effects whether or not the BIOS will fully configure all ISA-PnP cards. > 2) The PnP layer knows which resources are used by active devices, even > if they have not been allocated using e.g. request_region(). This > happens in practice. For example: Yes, this is an advantage. It does miss all the resources used by old ISA cards for which no (normal or pnp protocol as in this one) drivers have been loaded but I guess that's not much of a difference with a regular ISA driver... > 3) Knowing which resources a card can utilize, the PnP layer can try to > stay away from those resources when configuring other cards. The PnP layer > doesn't do that yet in a stock kernel, but I've patched it to do so. I'll > try to get those patches into the kernel. I guess that could be an advantage. When I was looking at pnp/isa integration a while ago myself I was looking at it from the angle of generic old ISA (ie, hardwired non settable resources) and concluded that for these there wasn't any point. Having looked a little closer at the resource assigment being done in the PnP layer now, I now feel there's some point, but still little for those. But with software settable devices such as this Jazz16, there ofcourse is rather more point. I believe the routing issue I refferred to above is the most important thing to have answered. I also still don't really like "splitting the one driver into the two pieces" like that but well, we don't actually have all that many software settable ISA devices and I guess it's not _too_ bad to have a few of these "front-ends" run as a protocol driver at boot as long as it stays "a few". Hey, I already said I like the concept... By the way; in the "straightforward version" I posted yesterday I neglected to reserve the SB_JAZZ16_CONFIG_PORT and the SB range before poking at it during setup. Attached version corrects that. Updated patch including your SB changes still at http://members.home.nl/rene.herman/jazz16.diff Looks more involved now, but posted mainly for completeness; correcting yesterday's post. Has been tested and found working again. Rene. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 9fef210..b60690a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1120,6 +1120,13 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based Media Vision Jazz16 chip + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/include/sound/sb.h b/include/sound/sb.h index 2dd5c8e..9594f89 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ }; #define SB_OPEN_PCM 0x01 @@ -140,8 +141,11 @@ #define SB_DSP_LO_OUTPUT_AUTO 0x1c #define SB_DSP_LO_INPUT_AUTO 0x2c #define SB_DSP_HI_OUTPUT_AUTO 0x90 #define SB_DSP_HI_INPUT_AUTO 0x98 +#define SB_DSP_ULAW_OUTPUT_AUTO 0x7f +#define SB_DSP_ULAW_INPUT_AUTO 0x8f #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +269,14 @@ #define SB_DMASETUP_DMA5 0x20 #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 + +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index a629b52..450255f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -258,6 +258,14 @@ config SND_SW60XG To compile this driver as a module, choose M here: the module will be called snd-sw60xg. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say 'Y' or 'M' to include support for Media Vision Jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index fd9d9c5..745e1c1 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 0000000..988a5e9 --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,334 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * The chip has an MPU-401 interface which is handled by the MPU-401 driver. + * + * The IBM PPS Model 6015 has a Jazz16 chip on board. Please tell me if it + * works with this driver or not. + */ + +#include <sound/driver.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/isa.h> +#include <linux/ioport.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +#define CRD_NAME "Media Vision Jazz16" +#define DEV_NAME "jazz16" + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_LICENSE("GPL"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +static long port = SNDRV_DEFAULT_PORT1; /* 0x220,0x240,0x260 */ +static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x310,0x320,0x330 */ +static int irq = SNDRV_DEFAULT_IRQ1; /* 5,9,3,7,10,15 */ +static int mpu_irq = SNDRV_DEFAULT_IRQ1; /* 5,9,3,7,10,15 */ +static int dma8 = SNDRV_DEFAULT_DMA1; /* 1,3 */ +static int dma16 = SNDRV_DEFAULT_DMA1; /* 5,7 */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); +module_param(port, long, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param(mpu_port, long, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param(irq, int, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param(mpu_irq, int, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param(dma8, int, 0444); +MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); +module_param(dma16, int, 0444); +MODULE_PARM_DESC(dma16, "16-bit DMA # for " CRD_NAME " driver."); + +static int __devinit snd_jazz16_match(struct device *dev, unsigned int n) +{ + if (!enable) + return 0; + + if (port == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + return 0; + } + if (irq == SNDRV_AUTO_IRQ) { + snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + return 0; + } + if (dma8 == SNDRV_AUTO_DMA) { + snd_printk(KERN_ERR "%s: please specify dma8\n", dev->bus_id); + return 0; + } + if (dma16 == SNDRV_AUTO_DMA) { + snd_printk(KERN_ERR "%s: please specify dma16\n", dev->bus_id); + return 0; + } + return 1; +} + +#define BUSY_LOOPS 100000 + +static int __devinit jazz16_sbdsp_command(u8 value) +{ + int i; + + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(port, STATUS)) & 0x80) == 0) { + outb(value, SBP1(port, COMMAND)); + return 0; + } + return -EIO; +} + +static int __devinit jazz16_sbdsp_get_byte(void) +{ + int i; + + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(port, DATA_AVAIL)) & 0x80) + return inb(SBP1(port, READ)); + + return -EIO; +} + +static int __devinit jazz16_sbdsp_reset(void) +{ + int error; + + outb(1, SBP1(port, RESET)); + udelay(10); + outb(0, SBP1(port, RESET)); + udelay(30); + + error = jazz16_sbdsp_get_byte(); + if (error < 0) + return error; + + return error == 0xaa ? 0 : -ENODEV; +} + +static void jazz16_set_ports(u8 value) +{ + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(value, SB_JAZZ16_CONFIG_PORT); +} + +#define JAZZ16_SET_DMAINTR 0xfb + +static int __devinit jazz16_enable(struct device *dev) +{ + static u8 irq_config[] = {0,0,2,3,0,1,0,4,0,2,5,0,0,0,0,6}; + static u8 dma_config[] = {0,1,0,2,0,3,0,4}; + u8 config = 0; + + int error; + + if (port != 0x220 && port != 0x240 && port != 0x260) + port = 0; + + if (mpu_port != 0x310 && mpu_port != 0x320 && mpu_port != 0x330) + mpu_port = 0; + + config = (port & 0x70) | ((mpu_port >> 4) & 0x07); + + snd_printd("%s: setting SB port = %#lx, MPU port = %#lx.\n", + dev->bus_id, port, mpu_port); + + jazz16_set_ports(config); + + snd_printd("%s: setting SB irq %d, dma %d&%d, MPU irq %d.\n", + dev->bus_id, irq, dma8, dma16, mpu_irq); + + if (!request_region(port, 16, "Jazz16")) + return -EBUSY; + + error = jazz16_sbdsp_command(JAZZ16_SET_DMAINTR); + if (error < 0) + goto out; + + config = dma_config[dma8] | dma_config[dma16] << 4; + + error = jazz16_sbdsp_command(config); + if (error < 0) + goto out; + + config = irq_config[irq] | irq_config[mpu_irq] << 4; + + error = jazz16_sbdsp_command(config); + if (error < 0) + goto out; + + error = jazz16_sbdsp_reset(); + if (error < 0) + goto out; + + error = jazz16_sbdsp_command(SB_DSP_GET_JAZZ_VERSION); + if (error < 0) + goto out; + + error = jazz16_sbdsp_get_byte(); + if (error < 0) + goto out; + + error = SB_VERSION_IS_JAZZ16(error) ? 0 : -ENODEV; + +out: release_region(port, 16); + return error; +} + +static irqreturn_t snd_jazz16_interrupt(int irq, void *dev_id) +{ + return snd_sb8dsp_interrupt(dev_id); +} + +static void snd_jazz16_free(struct snd_card *card) +{ + jazz16_set_ports(0); + release_and_free_resource(card->private_data); +} + +static int __devinit snd_jazz16_probe(struct device *dev, unsigned int n) +{ + struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int error; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (!card) + return -EINVAL; + + card->private_data = request_region(SB_JAZZ16_CONFIG_PORT, 1, + "Jazz16 Config"); + if (!card->private_data) { + error = -EBUSY; + goto out; + } + card->private_free = snd_jazz16_free; + + if (mpu_port == SNDRV_AUTO_PORT) + mpu_port = 0; + + if (mpu_irq == SNDRV_AUTO_IRQ) + mpu_irq = 0; + + error = jazz16_enable(dev); + if (error < 0) + goto out; + + error = snd_sbdsp_create(card, port, irq, snd_jazz16_interrupt, + dma8, dma16, SB_HW_AUTO, &chip); + if (error < 0) + goto out; + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(KERN_ERR "%s: not a Jazz16 chip at %#lx.\n", + dev->bus_id, chip->port); + error = -ENODEV; + goto out; + } + + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at %#lx, irq %d, dma8 %d, dma16 %d", + chip->name, chip->port, chip->irq, chip->dma8, chip->dma16); + + error = snd_sb8dsp_pcm(chip, 0, NULL); + if (error < 0) + goto out; + + error = snd_sbmixer_new(chip); + if (error < 0) + goto out; + + error = snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_AUTO, + 1, &opl3); + if (error < 0) + snd_printk(KERN_WARNING "%s: no OPL device found, skipping.\n", + dev->bus_id); + else { + error = snd_opl3_timer_new(opl3, 1, 2); + if (error < 0) + goto out; + + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } + + snd_card_set_dev(card, dev); + + error = snd_card_register(card); + if (error < 0) + goto out; + + dev_set_drvdata(dev, card); + return 0; + +out: snd_card_free(card); + return error; +} + +static int __devexit snd_jazz16_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __devinit alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, 1); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init (alsa_card_jazz16_init); +module_exit (alsa_card_jazz16_exit); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index aea9e5e..70cd031 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Sun Mar 11 18:45:38 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements. */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,14 +113,54 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + unsigned int num = 0, den = 0; + int err = 0; + + if (c->max > 1) + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + else + err = snd_interval_ratnum(r, 1, &clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -117,6 +169,12 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_MU_LAW) { + chip->playback_format = SB_DSP_ULAW_OUTPUT_AUTO; + break; + } + /* fall through */ case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; @@ -134,8 +192,26 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + switch (runtime->format) + { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_MU_LAW: + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + break; + + default: + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -173,7 +249,7 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -232,9 +308,11 @@ static int snd_sb8_capture_prepare(struc unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -242,6 +320,12 @@ static int snd_sb8_capture_prepare(struc chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } + /* fall through */ + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_MU_LAW) { + chip->capture_format = SB_DSP_ULAW_OUTPUT_AUTO; + break; + } chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO; break; case SB_HW_201: @@ -261,10 +345,26 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + switch (runtime->format) + { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_MU_LAW: + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + break; + + default: + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -278,14 +378,14 @@ static int snd_sb8_capture_prepare(struc } else { snd_sbdsp_command(chip, 256 - runtime->rate_den); } - if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->capture_format != SB_DSP_INPUT) { count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -358,10 +458,12 @@ static snd_pcm_uframes_t snd_sb8_playbac { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_PLAYBACK_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -369,10 +471,12 @@ static snd_pcm_uframes_t snd_sb8_capture { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_CAPTURE_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -443,6 +547,20 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_MU_LAW; + runtime->hw.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -463,9 +581,11 @@ static int snd_sb8_open(struct snd_pcm_s default: break; } + if (chip->hardware == SB_HW_JAZZ16) + return 0; snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); - return 0; + return 0; } static int snd_sb8_close(struct snd_pcm_substream *substream) diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3094f38..69e72b5 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 490b1ca..b16b592 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16:

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[alsa-devel] [Fwd: Re: [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards]
by Rene Herman 20 Mar '07

20 Mar '07

I didn't see this one make the old list at all, so forward to the new one... -------- Original Message -------- Subject: Re: [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards Date: Mon, 19 Mar 2007 23:46:07 +0100 From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> To: Rene Herman <rene.herman(a)gmail.com> CC: alsa-devel(a)lists.sourceforge.net, Adam Belay <ambx1(a)neo.rr.com> References: <20070311232436.GB9778(a)sygehus.dk> <45FDDA0F.7050503(a)gmail.com> On Mon, Mar 19, 2007 at 01:32:15AM +0100, Rene Herman wrote: > > I just now tested this on a standalone Jazz16 and it seems to be working > fine for me. I once had ogg123 segfault on me at the end of playback but > was unsuccesful in reproducing -- may have been a fluke. Thanks. I'll try out ogg123. > As to the comments... > > The changes to the SB code to support the Jazz16 I ofcourse agree with. > Supporting a new card is great. I am however not a huge fan of the new > PnP protocol. It's cute, but I feel it's a little over-engineered and > could just as well live directly in snd-jazz16. See below. > This setup also shares that same problem with the BIOS generally not > having reserved/routed the resources -- on a general PCI/ISA system the > user needs to go into the BIOS setup and reserve the resources the card > is (or will be, in this case) set to. So now the user really needs to > know what resources are going to be assigned, destroying the plug and > play idea... Not really. A standards compliant PnPBIOS set up with "PnP aware OS = YES" will use only enough resources to boot the system, which should leave enough irqs and dmas for the Linux PnP layer to do its work. I think there are three good reasons to use the PnP layer: 1) The PnP layer informs the PCI layer of irqs to try to avoid - please see pnp_register_irq_resource() in drivers/pnp/resource.c. 2) The PnP layer knows which resources are used by active devices, even if they have not been allocated using e.g. request_region(). This happens in practice. For example: $ cat /sys/bus/pnp/devices/01\:02.00/resources state = active io 0x2a0-0x2af io 0x390-0x393 io 0x500-0x50f irq 5 dma 0 dma 1 $ cat /proc/ioports [...] 0200-0200 : ns558-pnp 0201-0201 : ns558-pnp 0208-020f : ns558-pnp 0213-0213 : ISAPnP 0280-028f : ES18xx 0356-0356 : ide2 0360-0361 : MPU401 UART 0376-0376 : 0000:00:07.1 0376-0376 : ide1 0390-0391 : OPL2/3 (left) 0392-0393 : OPL2/3 (right) 03c0-03df : vga+ 03e8-03ef : serial 03f6-03f6 : 0000:00:07.1 03f6-03f6 : ide0 03f8-03ff : serial 0500-050f : AD1816A 0800-0807 : ES18xx - CTRL [...] Notice that 0x2a0-0x2af is missing from /proc/ioports. The PnP layer will avoid using that port range because it knows a device is using it. 3) Knowing which resources a card can utilize, the PnP layer can try to stay away from those resources when configuring other cards. The PnP layer doesn't do that yet in a stock kernel, but I've patched it to do so. I'll try to get those patches into the kernel. An advantage of autoprobing in a PnP protocol driver (as opposed to doing so in a sound card driver) is that it happens before any file systems have been mounted, so if the autoprobe happens to find one of those #%!$@ NE2000 cards and lock up the system, it won't corrupt your disks. -- Rask Ingemann Lambertsen

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[alsa-devel] test1
by Jaroslav Kysela 20 Mar '07

20 Mar '07

test1 ----- Jaroslav Kysela <perex(a)perex.cz>

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[alsa-devel] test
by Jaroslav Kysela 20 Mar '07

20 Mar '07

test ----- Jaroslav Kysela <perex(a)perex.cz>

1 0