From mboxrd@z Thu Jan 1 00:00:00 1970 From: bugtrack@alsa-project.org Subject: [ALSA - driver 0001611]: PCM to spdif blocked(any AC95), mic to spdif (in ALC650/655/850) Date: Wed, 7 Dec 2005 15:56:16 +0100 Message-ID: Mime-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit Return-path: Received: from bugtrack.alsa-project.org (gate.perex.cz [85.132.177.35]) by alsa.jcu.cz (ALSA's E-mail Delivery System) with ESMTP id 6E33C171 for ; Wed, 7 Dec 2005 15:56:16 +0100 (MET) Sender: alsa-devel-admin@lists.sourceforge.net Errors-To: alsa-devel-admin@lists.sourceforge.net List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , List-Archive: To: alsa-devel@alsa-project.org List-Id: alsa-devel@alsa-project.org A NOTE has been added to this issue. ====================================================================== ====================================================================== Reported By: psa Assigned To: ====================================================================== Project: ALSA - driver Issue ID: 1611 Category: PCI - via82xx Reproducibility: always Severity: major Priority: normal Status: new Distribution: Kernel Version: ====================================================================== Date Submitted: 12-02-2005 16:49 CET Last Modified: 12-07-2005 15:56 CET ====================================================================== Summary: PCM to spdif blocked(any AC95), mic to spdif (in ALC650/655/850) Description: After transition 1.0.9b -> 1.0.10 PCM output (or 'Analig In', if ADC -> spdif supported) can't be directed to spdif output. Caused by a feature bug in via82xx-driver. Driver now enables spdif out only if IEC958 Playback AC97-SPSA dxs3 (slot 10/11 reserved for non PCM like ac3) is selected. This is obviously done with the assumtion that if user program wants to send something to spdif it uses dxs3 and/or with the assumption that only non PCM strams can be directed to spdif port. However both of these assumtions are too restrictive. Even further these assumprions ignore the fact that several ac97 chips (like ALC650) can direct ADC output (originaly intended for capture) to spdif output; and some chips (e.g. ALC655/850) can direct spdif input directly to spdif output. However many ordinary sound programs (like internet radio) have no control on which dxs they direct their output. They put PCM (stereo or prologic) to standard stereo output which is dxs0 (slot 3/4). In the other end of the spdif cable there can be all kind of hardware, including amplifiers that can take several formats including standard PCM. In makes perfect sense to direct dxs0 to such spdif devices. Also since, although slots have certain standard usages, user software is free to use them as they wish, there is no reason to restrict the slot selection. In addition dxs selection is of no relevance when spdif out signal is taken from ADC or spdif input. Yet it should be possible to enable spdif output. The first restriction is implemented in the code in pci/via82xx.c in snd_via8233_playback_prepare(): if (chip->spdif_on && viadev->reg_offset == 0x30) snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate); where reg_offset 0x30 is the offset of DXS3. snd_ac97_set_rate() again is the place (only place as far as I can see) spdif output is enabled. The second restriction is implemeted by the fact that 'IEC958 Output'-swicth does not enable spdif output. It only sets '->spdif_on'. This all means that spdif output is anabled only if user software is started with spdif_on set (by 'IEC958 Output'-swicth). E.g can't do the following: Set 'Capture Source' select to Mic, redirect ADC (Analog In) to spdif output, set 'IEC958 Output'-swicth (, select DXS0). Now you should here microphone trouhg you external amplifier connected by spdif even if no software is running. No can do. Suggested fix: The first restriction is simply removed by removin the second condition in the if-statement: if (chip->spdif_on) snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate); The second restriction is removed by adding to snd_via8233_dxs3_spdif_put() enbaling/disabling spdif output. In addition it suggested that in case ac97 chip supports directing ADC to spdif output (like ALC650/655/850), in pci/ac97/ac97_patch.c in appropriate *_patch a ADC source select is added on the playback side (kind of duplicating 'Capture Source') since this select is used also to select spdif output in case ADC is directed to spdif output, and is thus a 'playback'-mixer-control. Diffs are included. Note that snd_via8233_dxs3_spdif_*:s have been changed to snd_via8233_spdif_*:s since they are no more specific to dxs3. There is also a diff for adding the mixer control descriped above for ALC850 as an exsample. In setting available for me everything seems to work fine above restrictions removed. via82xx diff enclosed. ac97_patch diff: ----------------------------------------------------------------------------- *** pci/ac97/ac97_patch.c.myorg Thu Nov 24 19:22:05 2005 --- pci/ac97/ac97_patch.c Sun Nov 27 23:22:48 2005 *************** *** 2106,2111 **** --- 2106,2115 ---- 0); } + static const char *alc655_rec_sel[] = { "Mic", "CD", "Mute", "Aux", "Line", "Mix", "Mono Mix", "Phone"}; + + static const struct ac97_enum alc655_enum = AC97_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, alc655_rec_sel); + static const snd_kcontrol_new_t snd_ac97_spdif_controls_alc655[] = { AC97_PAGE_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_ALC650_MULTICH, 11, 1, 0, 0), /* disable this controls since it doesn't work as expected */ *************** *** 2117,2122 **** --- 2121,2127 ---- .get = alc655_iec958_route_get, .put = alc655_iec958_route_put, }, + AC97_ENUM(SNDRV_CTL_NAME_IEC958("Analog ",PLAYBACK,NONE), alc655_enum), }; static int patch_alc655_specific(ac97_t * ac97) --------------------------------------------------------------------------- ====================================================================== ---------------------------------------------------------------------- psa - 12-06-05 03:28 ---------------------------------------------------------------------- > 2: The ALSA PCM name refers to both the route and its content. > For example, "spdif" means the SPDIF I/O and its accepted h/w configuration > (e.g. format, channels - in most cases S16, 2 channel). Ditto for > "surround51". It refers to the 5.1 output and its accepted h/w format. So, > it's not conceptually mutual. Aren't you hard headed. Now tell me please from where do get your argument. By whos definition SPDIF implies format or slot? It implies only route, nothing else. Havn't you read the AC97 specs or IEC958 specs? It is really the contrary to what you say: PCM implies format, SPDIF does not; it allows all kind of formats and contents; it's just a carrier. As to the "surround51", it is quite ambiguous as well. It implies content but not format. It can be 6 channel (3 * PCM slot), AC3 or DTS at least. But since that call is snd_pcm_open() I assume it means 3 * PCM. How do you open AC3 stream? I can well see that it must be determined befor the driver whether to use software AC3-decoder or put it trough as such to driver. But even if it is put to driver it does not yet imply SPDIF. There could be a hardware decoder on the controller or codec chip (not AC97 though). Then again it would be the driver to determine what to do with the stream. There again it would make sense to have a mixer switch to let the human user to determine final destination. In my opinion it is for sound producing software to determine the final output only if the format and content allows no alternatives or if the software has a human interface to set it. Other wise it must be left to the mixer. Standard PCM allows alternatives, so why not to leave it to mixer to route? > It's software who determines the output to use, indeed, from the system > viewpoint. We can make the arbitrary output and change the output route > dynamically, but it's a different question. This is not what the kernel driver > code is involved. That is, your approach isn't a good solution. If it is for sound producing software to determine the final destination of its product, then what are the mixer software for? Indeed what do you think there are any sound mixers in the world, if not for 'the arbitrary output and change the output route dynamically'. And if that is not what the kernel driver code is involved with, then how can the mixer inside the codec chip be controlled by the user. Isn't the alsamixer there exactly for doing that trough kernel driver. Are you really saying, that you have decided, that it is wrong to listen arbitary sound program, that produces standard stereo PCM, trough my amplifier that is connected to my computer by SPDIF-cable, even though it works in practice and comforms with AC97 specs? And how would it contradict the driver design? It seems to me that the driver and controls there are designed exactly for that. If not, for what? And what is the alsamixer for? I would hate to extend this discussion to more complex mixers (like SB Live!) that has several SPDIF inputs and outputs (and other inputs and outputs of all kinds). How on earth could the sound producing software know where the sound is supposed to end up. It would be possible only if there would be fully fledged mixer as part of every sound producing software. Conceptually mixer is a device that can take input (in this case sound input) from various different sources in various formats as the human user determines mix them together and route them (and maybe translate their format) to various output connectors in various formats as determined by human user. Why pretend that there isn't such capability (albeit limited) in the hardware (that's what the harware designer has intended it for). If you have not decided that it is wrong to listen simple internet radio program or simple CD-player program, that has not a comprehensive mixer as part of it, trough a spdif cable, then what is your sollution? -> 4: Your patch doesn't provide the proper notification to multiple controls, -> so the race occurs very easily on a single app. As said, having two -> identical controls with different names is simply bad. Please don't. Ok, mayby it's bad. But then other sollution must be found. I just wonder why the notification goes only one way. The code for both in the driver is exactly the same. Only difference is the name: "Capture Source" vs. "Playback Analog Source". It must be in the actual alsamixer code since the latter control appears as standard two channel enum control while the former has special format of display. My guess is, the capture side code of the mixer does not react to notification since it has different name, while the standard enum code somehow recognizes that this is really same control just with different name. Any way alternate sollution would be to change alsamixer code. But how could it know if the codec chip has this capability? -> 5: I don't understand your statement. If the codec chip can't set the proper -> IEC958 status bits for the given sample rate, you can't use it. (And, if ADC -> is linked to SPDIF, why bother to change the PCM code?) Ah, now I get it. Are you talking about the 'Validity bits' in MAX2A and MX3A? It would be a good idea to use them. But current code does not. Instead it just puts max and min valus for hw.rate inside snd_pcm_limit_hw_rates(). No checking of validity bits, at least not here. if (chip->spdif_on && viadev->reg_offset == 0x30) { /* DXS#3 and spdif is on */ runtime->hw.rates = chip->ac97->rates[AC97_RATES_SPDIF]; snd_pcm_limit_hw_rates(runtime); The exsample situation I was descriping would not require setting limits to AC-link/DAC rate, as this code does. I.e code does not allow AC-link/DAC run 44.1 and ADC/SPDIF-transmitter 48, but forces both to run 48. ---------------------------------------------------------------------- tiwai - 12-07-05 15:56 ---------------------------------------------------------------------- Please read my comment carefully before emotional reaction. "spdif" PCM implies the I/O from/to SPDIF _and_ the formats accepted by _hardware_. In your case, it's S16_LE 2channels. It's quite hardware specific, not per SPDIF definition. Ditto for "surround51". It routes to analog 5.1 outputs _and_ restricts the hardware configuration that the hardware accepts. OK? (Also, I didn't mention about slots anywhere.) It's fine to allow the SPDIF-output with the "default" PCM output. But, it's not mandatory. Instead, "spdif" PCM assures to output to SPDIF even with non-audio data. For the "default" PCM, usually the concurrent access/mix is allowed, so it's not suitable for non-audio data. About the mixer: You think obviously that the sound system consists of only a kernel driver. No, we have a user-space layer to mangle many things. The mixer is indeed a part of alsa-lib stuff. About the SPDIF output sample rate: It's PRSR bits of register 3A. You'll find that only limited sample rates are available for SPIDF out with AC97 codecs. Anyway, the further argument is pretty useless. Let's stop it. If you want to continue, please post it to alsa-devel ML instead of the bug report. Now, let's revise the possible improvements. For the SPDIF output on VIA823x, we have the following capabilities: - DXS3 dedicated for SPDIF The merit is that you can play still 3 DXS analog streams independently without signal mix. That is, you can avoid dmix and still use SPDIF safely even with non-audio data. The demerit is that DXS output is fixed to 48kHz (it does SRC by itself). (So, basically, it's a bug to call snd_ac97_set_rate(AC97_SPDIF) in snd_via8233_playback_rate. It has to be fixed...) We can set SPDIF slots 3&4 as default for the concurrent output, and disable DXS3 unless the "spdif" PCM is opened. The slot assigment can be then changed automatically to 10&11 when "spdif" PCM is opened (via VIA8233.conf). - Multi-channel stream for both analog and SPDIF with slot 3&4 The merit is that it allows you non-48k (32k and 44.1k), too. However, multi-channel stream is exclusive. That is, no DXS is possible. In addition, we'll need a mechanism to shut up the analog output automatically in the case of non-audio data by checking the SPDIF status bits. To my eyes, the first one looks simplier. Issue History Date Modified Username Field Change ====================================================================== 12-02-05 16:49 psa New Issue 12-02-05 16:49 psa File Added: via82xx.c.diff 12-02-05 16:51 psa File Added: ac97_patch.c.diff 12-02-05 17:39 tiwai Note Added: 0006897 12-03-05 19:07 psa File Added: via82xx.c.udiff 12-03-05 19:08 psa File Added: ac97_patch.c.udiff 12-03-05 21:18 psa Note Added: 0006903 12-05-05 12:02 tiwai Note Added: 0006910 12-05-05 21:15 psa Note Added: 0006916 12-05-05 21:45 tiwai Note Added: 0006917 12-06-05 03:28 psa Note Added: 0006919 12-07-05 15:56 tiwai Note Added: 0006943 ====================================================================== ------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. 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