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[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

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[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-

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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 05 Apr '07

05 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-----

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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: > /* > * 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-----

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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.

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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

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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

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[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

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[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

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[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.

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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

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[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

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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

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[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

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[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.

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[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. ********************************

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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

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[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.

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[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

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[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); }

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[alsa-devel] [2/4] 2.6.21-rc5: known regressions (v2)
by Adrian Bunk 31 Mar '07

31 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

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[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 27 Mar '07

27 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

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[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

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[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. */

4 5

[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

1 1

[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; }

1 0

[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

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[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!

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[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/

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[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.

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[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. ********************************

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[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

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[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

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[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>

1 0

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

20 Mar '07

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

1 0