Sunday, March 20th, 2011, 00:06 UTC | ||
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[00:07:33] | knightr: | !seen reynaldo |
[00:07:33] | MythLogBot: | reynaldo is here and has been idle for 15 days 7 hours 8 minutes 50 seconds |
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[10:28:08] | maccco: | http://www.mythtv.org/wiki/User_Manual:HDAudioPassthrough it says nvidia gforce 8xxx doesn't support hbr. But i'm able to send 8 discreate 192khz channels to my recevier |
[10:28:24] | maccco: | i hame a mcp78 (gforce 8300) |
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[23:02:19] | Saviq: | hi all, do you think it would be possible / difficult to make mythtranscode pass untouched audio in fifo mode? |
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[23:10:10] | wagnerrp: | might want to talk to beirdo about that |
[23:10:22] | wagnerrp: | he was the last one to touch that stuff, used for nuvexport |
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[23:20:52] | Beirdo: | not currently possible |
[23:21:21] | Beirdo: | and I think jya was going to look at options for changing it, but not sure how far he got, etc. |
[23:21:53] | Beirdo: | in fifo mode you get PCM output (with as many output channels as there were in the source) |
[23:22:19] | jya: | Beirdo: I stopped looking at the AC3 audio support after realising it would require changing the nuv container code. |
[23:22:29] | Beirdo: | ah yeah |
[23:22:33] | jya: | which from everyone's opinion is something we should get rid of soon |
[23:22:56] | wagnerrp: | fair assumption |
[23:23:07] | Beirdo: | aye. |
[23:23:20] | jya: | I just didn't see the point on working on something that would be obsolete before it's even official |
[23:23:21] | Saviq: | jya: what about fifos? how difficult do you think it would be for fifos to spit out untouched audio? |
[23:24:12] | jya: | Saviq: to spit out untouched audio, you need to be able to put in the container what audio codec it is.. and nuv supports only a very very limited subset of audio codec |
[23:24:49] | jya: | no point putting a DTS stream for example, if the player has no way of knowing that it is a dts stream |
[23:25:00] | Beirdo: | in fifo mode, there is no container. it's uncompressed video frames and PCM audio |
[23:25:07] | Saviq: | yup, that ^ |
[23:25:11] | Beirdo: | on two separate fifos |
[23:25:19] | jya: | Beirdo: but then that's not *untouched* audio |
[23:25:26] | Saviq: | yes |
[23:25:30] | Beirdo: | agreed |
[23:25:34] | jya: | or we have a different definition of what's untouched audio is |
[23:25:38] | Saviq: | and that's something I'd like to change |
[23:25:50] | Saviq: | instead of uncompressing the audio to PCM |
[23:25:57] | Saviq: | just keep it in the original format |
[23:26:04] | jya: | Saviq: and for that you need to change the nuv container |
[23:26:05] | wagnerrp: | you want to get the mp3 out? |
[23:26:06] | Beirdo: | use ffmpeg |
[23:26:10] | jya: | it only supports pcm and mp3 audio |
[23:26:13] | Beirdo: | or even mythffmpeg |
[23:26:21] | Saviq: | wagnerrp: mp3 / dts / ac3 / whatever |
[23:26:27] | wagnerrp: | no, only mp3 (or pcm) |
[23:26:38] | Beirdo: | if you are coming from nuv, you are hosed, but that's only used for framegrabbers |
[23:26:48] | jya: | Saviq: the nuv we generate is already a myth-only thing, mplayer or VLC would most likely be unable to play the file |
[23:26:50] | wagnerrp: | the only reason to use mythtranscode's pipes are because you dont support nuv natively |
[23:26:57] | wagnerrp: | if its another format, like mpg |
[23:27:01] | jya: | adding support to other audio types is going to break that even further |
[23:27:01] | Beirdo: | if you are recording MPEG2 or H.264, you are already in MPEG2-TS or -PS container |
[23:27:09] | wagnerrp: | theres no reason why the transcoding software youre using cannot access that container directly |
[23:27:35] | Saviq: | I'm in mpeg2/4 land, so that's fine |
[23:27:48] | Saviq: | the only reason I wanted to use mythtranscode are cutlists |
[23:27:58] | Beirdo: | yeah |
[23:28:16] | wagnerrp: | does the pipe mode apply the cutlist? |
[23:28:22] | Beirdo: | well, it's something that will need attention, but I don't think it will be any time soon |
[23:28:24] | jya: | Beirdo: did multi-channels audio in mythtranscode ever worked? |
[23:28:28] | Beirdo: | yes, fifo mode does cut list |
[23:28:40] | Beirdo: | jya: works fine |
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[23:28:53] | jya: | Beirdo: what's this issue of it playing too fast then? |
[23:28:55] | Beirdo: | it spits out as many channels as the source was |
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[23:29:34] | Saviq: | so... all in all I could just reencode audio, too |
[23:29:34] | Beirdo: | that's because it's putting out 6 channels when the consumer is expecting 2... or other variations |
[23:29:41] | Beirdo: | AFAIK |
[23:29:54] | Saviq: | might be the sane thing to do here |
[23:30:15] | jya: | but if there are 6 channels audio in the data, the container should have 6 as the number of channels |
[23:30:26] | Saviq: | jya: there's no container on the fifo |
[23:30:28] | jya: | so if you playback in stereo or 6 channels it will make no difference |
[23:30:37] | Beirdo: | what container? |
[23:30:45] | jya: | Saviq: isn't it a nuv file? |
[23:30:46] | jya: | I'm lost |
[23:30:46] | Saviq: | jya: fifo |
[23:30:47] | Beirdo: | in fifo mode there IS NO CONTAINER |
[23:30:51] | Saviq: | it's a named pipe |
[23:30:57] | jya: | ah... |
[23:30:58] | Beirdo: | you get a fifo of video frames |
[23:30:59] | Saviq: | with mythtranscode --fifodir |
[23:31:05] | Beirdo: | and a fifo of PCM samples |
[23:31:09] | jya: | Saviq: what's the point of fifo ? |
[23:31:19] | wagnerrp: | for stuff that doesnt support nuv containers |
[23:31:32] | Beirdo: | to use with nuvexport, etc with cutlist support |
[23:31:51] | Beirdo: | and for nuv source and external encoders |
[23:31:52] | jya: | is there any ways to pass the number of channels used? how did this ever worked then? |
[23:31:53] | Saviq: | jya: with fifo you get uncompressed, nicely cut raw video and audio |
[23:32:08] | Saviq: | jya: that you need to know in advance |
[23:32:27] | Beirdo: | we probe the original file using mplayer (if non-nuv) |
[23:32:40] | jya: | Saviq: I still don't understand how this ever worked. If you put 6 channels of lpcm audio, and the player thinks it's two. How did this ever worked |
[23:32:45] | Beirdo: | if nuv, we read the nuv container |
[23:33:02] | Saviq: | jya: you don't, you need to pass the player the format |
[23:33:09] | Saviq: | player / encoder |
[23:33:14] | jya: | Beirdo: ah, so I guess for AC3 , it looks at the snumber of channels and not the number of ac3 channels (2 vs 6) |
[23:33:17] | Saviq: | same for video frames |
[23:33:47] | Beirdo: | no, it decodes out to 2 or 6 PCM channels, and the consumer has to know which |
[23:34:03] | Beirdo: | which it currently can get from mplayer -identify |
[23:34:16] | jya: | thinking more about i ; if you pass the right data, I still don't understand why it wouldn't work, and if it doesn't work now why it would have worked before. nothing fundamentally has changed here |
[23:34:29] | Saviq: | jya: nothing's broken |
[23:34:31] | Beirdo: | which is why I added code in nuvexport to downmix the 6 channels to 2 channels using sox |
[23:34:53] | jya: | so why are people complaining that it plays too fast (or too slow) |
[23:35:06] | ** jya very confused... ** | |
[23:35:07] | Saviq: | jya: I was just asking how difficult would it be to put AC3 frames instead of PCM on the fifo |
[23:35:24] | Beirdo: | because they won't give us debugging information to determine their problems. |
[23:35:34] | jya: | Saviq: and my answer to that is that you need to extend the nuv container to support other audio codecs. |
[23:35:42] | Saviq: | ouch |
[23:35:49] | Beirdo: | it's likely they are getting 2 channels from mythtranscode and thinking they have 7 |
[23:35:49] | jya: | Beirdo: so that too fast/slow can be ignore for now ? |
[23:35:50] | Saviq: | there's no container on the fifo to be extended |
[23:36:12] | Beirdo: | nuv has nothing to do with fifo mode |
[23:36:19] | jya: | Saviq: sorry I misread |
[23:36:42] | Beirdo: | err 6 rather than 7 above... mistype |
[23:36:49] | jya: | Saviq: avfd used to output an iec958 encoded frame when you played an AC3 or DTS track |
[23:36:59] | jya: | so you could never get the native, raw audio stream |
[23:37:16] | Beirdo: | what we need to be able to debug these issues is to get a sample recording, and find out all the commandlines they are trying to use |
[23:37:26] | jya: | now avformatdecoder output the raw audio stream, and it's up to the audio class to encapsulate the data in a iec958 frame |
[23:37:59] | Beirdo: | I'd be worried about maintaining sync on the fifos if you keep the original audio, but I'm sure there's a decent way to spit them out at the right rates |
[23:38:29] | Saviq: | jya: so the dummy audioplayer in transcode.cpp could be tweaked to "support passthrough" |
[23:38:53] | wagnerrp: | honestly, i would just do a 2-stage transcode |
[23:38:57] | jya: | Saviq: as such, if the fifo mode use avformatdecoder to decode the data ; it will get raw audio if it is configured to output raw audio (passthru flag set) ; from there it's up to the audio code to handle it ; which with mythtranscode is its own audio class |
[23:39:02] | wagnerrp: | do lossless through mythtranscode directly |
[23:39:08] | wagnerrp: | and then do your own thing from there |
[23:39:27] | Saviq: | wagnerrp: or simply recompress audio, too... |
[23:39:31] | jya: | Saviq: what class does transcode.cpp uses to decode audio and video ? |
[23:39:47] | Saviq: | jya: MythPlayer ;) |
[23:40:33] | Saviq: | https://github.com/MythTV/mythtv/blob/master/ . . . ode.cpp#L385 |
[23:40:58] | Saviq: | it connects a dummy AudioOutput device https://github.com/MythTV/mythtv/blob/master/ . . . code.cpp#L36 |
[23:41:02] | Beirdo: | getting the cutlist, etc right with passthru of original audio and keeping sync would be the two main challenges, I'd expect |
[23:41:06] | jya: | ok, so that's using avfd |
[23:41:24] | jya: | it's just a matter of setting the passthru code then, what addframe will then receive is raw audio |
[23:41:41] | Saviq: | cool |
[23:41:52] | jya: | there are methods that needs to be overriden so avfd knows that it can do passthru |
[23:42:13] | jya: | I never bothered to re-implement those classes for mythtranscode |
[23:42:34] | jya: | personally, I wouldn't have done it that way: using a dummy audio class |
[23:43:21] | jya: | it's going to be a lot of work to properly support it. I would have instead written a low-level audio drive (similar to the alsa, pulse, null) that would write to a buffer |
[23:43:50] | jya: | because right now, that dummy audio class is a higher level one (same as audiooutputbase) that's a lot of code to rewrite |
[23:45:08] | jya: | Saviq: is that something you want to do ? or you're waiting for someone else to do it ? |
[23:45:24] | jya: | problem is that I have *never* used fifo mode ; so I wouldn't know how to even use the command line |
[23:46:11] | Saviq is now known as Saviq_afk | |
[23:46:57] | Beirdo: | if you need different behavior on the fifos, please add another command-line arg for the new behavior and leave the default as is. |
[23:47:06] | Beirdo: | to reduce other breakage :) |
[23:47:35] | jya: | I think if we simply added a --rawaudio ; and re-implement the CanPassthrough method to return "true" all the time |
[23:47:41] | jya: | then you get your raw audio |
[23:47:48] | jya: | (that's on master only) |
[23:48:13] | Beirdo: | I'd shy away from using hte word raw. |
[23:48:31] | Beirdo: | it's confusing in its connotation |
[23:48:33] | jya: | what would you call it ? |
[23:48:49] | Beirdo: | --originalaudio perhaps? |
[23:48:58] | jya: | (not that I actually was serious about a --rawaudio) |
[23:49:01] | Beirdo: | as original as we have it anyways |
[23:49:13] | jya: | was just saying what needs to be done to have "originalaudio" support |
[23:49:26] | Beirdo: | aye |
[23:49:29] | jya: | quite trivial when I think of it |
[23:50:21] | jya: | now that I've moved the iec958 encapsulation away from avformatdecoder.cpp ; there's no problem doing so |
[23:50:50] | jya: | I only moved the encapsulation out from avfd because Ithought it was more elegant from a design point of view |
[23:51:00] | jya: | but it's going to make things much easier for the use in mythtranscode |
[23:51:03] | Beirdo: | that whole thing needs a rework at some point of course (the transcode part) |
[23:51:20] | jya: | Beirdo: but with fifo, how do you maintain the timecode sync ? |
[23:51:47] | Beirdo: | the transcode code does it |
[23:51:55] | Beirdo: | or so goes the theory |
[23:52:14] | Beirdo: | it will only fill the fifos to keep it in sync (unless you tell it not to) |
[23:52:19] | jya: | but if you pass that fifo stream out and it only has the video & the audio |
[23:52:24] | jya: | how is sync maintained? |
[23:52:41] | Beirdo: | it has one fifo for video, one for audio |
[23:52:49] | jya: | especially with raw ac3 or dts, the packet size varies constantly |
[23:52:55] | Beirdo: | and it fills them at such a rate to keep em in sync |
[23:53:07] | Beirdo: | which is why I say it will become challenging |
[23:53:30] | jya: | with raw ac3/dts you can't base the rate on the packet size (which is what avfd assumes most of the time) |
[23:57:50] | myco (myco!~mycosys@proxy-prod1.newcastle.edu.au) has joined #mythtv | |
[23:58:43] | myco (myco!~mycosys@proxy-prod1.newcastle.edu.au) has quit (Remote host closed the connection) | |
[23:59:01] | jya: | Beirdo: looking at transcode.cpp ; it also assumes that the length of the audio data to be at a constant rate |
[23:59:09] | jya: | that won't work with raw ac3/dts |
[23:59:17] | myco (myco!~mycosys@proxy-prod1.newcastle.edu.au) has joined #mythtv | |
[23:59:33] | jya: | can fix that floatting point exception first.. why is eff_audiorate 0 ? |
[23:59:36] | Saviq_afk is now known as Saviq |
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