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Sujao
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PostPosted: Mon Jan 17, 2005 5:29 am    Post subject: howto convert ac3 to ogg? Reply with quote

Well the topic says it already. I am looking for a guide or a HOWTO on how to convert ac3 into ogg. I need to retain the 5.1 sound. Afaik OGG is capable of 5.1, is it? I searched the forums but didnt find a single thread about it.

I looked at doom9.org and vorbis.com and didnt find a signe article, what I find strange. I hope what I am looking for is possible at all.
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GordSki
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PostPosted: Mon Jan 17, 2005 1:54 pm    Post subject: Reply with quote

Found this: http://www.divx-deux.com/oggyaac/Manual%20OGG%20y%20AAC.htm

It's not in english and the progs are windows based, but it's a start......


G.
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iarwain
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PostPosted: Mon Jan 17, 2005 1:55 pm    Post subject: Reply with quote

HeadAC3He is a very good tool to do that. Yes, I know it's for windows. But read this, taken from the official web page:
Code:
... Even if some rumors floating around about me dicontinuing HeadAC3he, that is not completely true. There will be another release, which then probably will be the last. Not mainly because of lack of free time, but mostly due to tha fact I nearly completely switched to Linux, which is much better than Windows. For example: I recently bought a NuTech 8x DVD+RW burner. Burning under Windows makes my system slooow and unresponsive (I have a AthonXP@2,1GHz and 1GB RAM...and yes, DMA is activated), but in Linux the system doesn't even feel like doing something when burning (kernel 2.6.x, ATAPI). I LOVE my Linux... So whenever I decide to continue releasing HeadAC3he after next release, it will be a native Linux prog. As I probably will use gtk+ or wxWindows, it will also run natively in Windows. But this is no promise. It is just "If ... then ...".
BTW, HeadAC3he runs in a Linux WINE environment, though there are some issues with the windows, which I may resolve for the next release...

I hope he makes it.
With oggenc you can pass the "-C n" argument, where n is the number of channels. First you have to do the ac3->wav conversion (to 5.1 wav).
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Kraymer
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PostPosted: Mon Jan 17, 2005 4:53 pm    Post subject: transcode or mplayer Reply with quote

transcode should be able to do it:
Code:
transcode -i input.ac3 -y null,ogg="-q $quality" -N 0xfffe -m output.ogg
where you can set a quality level for oggenc.
I you don't have transcode it might be overkill to install, so have a look at mplayer:
Code:
mplayer -ao pcm input.ac3
oggenc -o output.ogg -q $quality audiodump.wav

Hope that gives you the right direction :)

Sebastian
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Sujao
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PostPosted: Tue Jan 18, 2005 6:49 am    Post subject: Reply with quote

yesterday I spent about 5 hours playing around with mlpayer, oggenc and transcode and got stuck at the step where I encode either a 6-channel-pcm or 6-channel-raw-stream-ac3 to ogg.

1. alternative
  • mplayer dumps the audiostream to a raw ac3 stream
    Code:
    mplayer -aid 128 -dumpaudio -dumpfile dumpfile /home/user/tmp/city/DVDVOLUME/VIDEO_TS/VTS_01_5.VOB


  • i can play it just fine with:
    Code:
    mplayer -rawaudio on:format=0x2000 dumpfile


  • then i use transcode to encode it to ogg. this way it works, but i only retain 2 channels
    Code:
     transcode -i dumpfile -x null,ac3 -y null,ogg="-q 4" -N 0xfffe -m transcoderesult.ogg


  • i added "-C 6" to the ogg options hoping to retain all 6 channels....
    Code:
    transcode -i dumpfile -x null,ac3 -y null,ogg="-q 4 -C 6" -N 0xfffe -m transcoderesult.ogg


but here I get the problem I was unable to solve. The resulting ogg is played to fast, about 30% too fast.

2. alternative
Before this I tried another way to get my audio stream and dumpled it directly to PCM with mplayer with this command:
Code:
mplayer -aid 128 -vc dummy -vo null -ao pcm waveheader /home/user/tmp/city/DVDVOLUME/VIDEO_TS/VTS_01_1.VOB

which worked too. but I got the same problem, after encoding to ogg. either I had only 2 channels or the audio played to fast.

Conclusion

So the problem seems to be OGG. Moreover I dont have a 5.1 sound system so I cant check if the 6 channels are present at all (any tipps on how to control this?)

So what is the problem with OGG? Any ideas how I could encode either ac3 or PCM and keep 6 channels and normal speed?


Additional Info

Here is the output of mplayer when I play the dumped raw-ac3-stream
Code:
...
Playing dumpfile.
==========================================================================
Opening audio decoder: [liba52] AC3 decoding with liba52
Using 3DNowEx optimized IMDCT transform
AC3: 5.1 (3f+2r+lfe)  48000 Hz  448.0 kbit/s
Using MMX optimized resampler
AUDIO: 48000 Hz, 2 ch, 16 bit (0x10), ratio: 56000->192000 (448.0 kbit)
Selected audio codec: [a52] afm:liba52 (AC3-liba52)
==========================================================================
Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...
AF_pre: af format: 2 bps, 2 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 2ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...
Video: no video
Starting playback...
...

Here is the output of mplayer when I play the 2-channel-ogg-stream
Code:
...
Playing transcoderesult2ch.ogg.
Cache fill: 15.04% (1261568 bytes)    Ogg file format detected.
==========================================================================
Opening audio decoder: [libvorbis] Ogg/Vorbis audio decoder
AUDIO: 48000 Hz, 2 ch, 16 bit (0x10), ratio: 16000->192000 (128.0 kbit)
Selected audio codec: [vorbis] afm:libvorbis (OggVorbis Audio Decoder)
==========================================================================
Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...
AF_pre: af format: 2 bps, 2 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 2ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...
Video: no video
Starting playback...
...

Here is the output of mplayer when I play the 6-channel-ogg-stream, that plays too fast
Code:
...
Playing transcoderesult6ch.ogg.
Ogg file format detected.
==========================================================================
Opening audio decoder: [libvorbis] Ogg/Vorbis audio decoder
AUDIO: 48000 Hz, 6 ch, 16 bit (0x10), ratio: 64500->576000 (516.0 kbit)
Selected audio codec: [vorbis] afm:libvorbis (OggVorbis Audio Decoder)
==========================================================================
Checking audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/2ch/16bit...
AF_pre: af format: 2 bps, 6 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 6ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/2ch/16bit...
Video: no video
Starting playback...
...
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Kraymer
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PostPosted: Tue Jan 18, 2005 7:56 am    Post subject: öööhhh Reply with quote

Sujao wrote:
Code:
 transcode -i dumpfile -x null,ac3 -y null,ogg="-q 4" -N 0xfffe -m transcoderesult.ogg
but here I get the problem I was unable to solve. The resulting ogg is played to fast, about 30% too fast.

hm okay, I really overlooked that you aimed to retain 5.1 sound. I wouldn't have suggested transcode then. the ac3-decoder will always (afaik) merge channels down to 2 because 2-channel-PCM is the basis for all the audio output modules..
Quote:

So what is the problem with OGG? Any ideas how I could encode either ac3 or PCM and keep 6 channels and normal speed?

Hm, the manpage indeed says that several channels are possible and that 2 are only a default.. I never heard of 5.1 sound in ogg though :(

Could it be that pcm only supports 2 channels? if not, maybe there is some console program that can output to 6-channel-pcm (to stdin so you can feed oggenc with that).

Sorry, I'm a bit lost on 5.1 ;)

Sebastian
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Cintra
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PostPosted: Tue Jan 18, 2005 9:06 am    Post subject: Reply with quote

ref http://www.un4seen.com/ it would seem multi-channel ogg is possible. Haven't played the sample yet myself tho'.

mvh

Edit: just played it but the outputs do not correspond to my speaker layout, and back left is missing, whereas DTS samples from http://www.sr.se/multikanal/english/e_index.stm do play from the correct speakers..
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GordSki
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PostPosted: Tue Jan 18, 2005 11:02 am    Post subject: Reply with quote

The reason it sounds faster is because it is treating a 2 channel input as a 6 channel input giving you a thrid of the samples you should have. So the oggenc part of the scheme is doing what it should be, it the input that is wrong.

Do you have an mplayer output for playing the PCM file you created? I suspect that mplayer has down mixed it to 2 channels before dumping the PCM.....


G.
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GordSki
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PostPosted: Tue Jan 18, 2005 11:19 am    Post subject: Reply with quote

After a quick bit of googling I can't seem to find any ac3 to 6-channel OGG or PCM linux progs......

Possibly your best bet is to try the CLI version of BeSweet under wine (http://dspguru.notrace.dk/).

Either that or get down and dirty with the mplayer code to remove the down mix 8O

Good luck!

G.
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GordSki
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PostPosted: Tue Jan 18, 2005 11:29 am    Post subject: Reply with quote

I think I've got it!

I've found this:

Quote:

mkfifo audiodump.wav
faac audiodump.wav test.aac & mplayer dvd://1 -vo null -vc null -ao pcm -channels 6 >/dev/null
rm audiodump.wav


on another forum (http://forum.doom9.org/showthread.php?s=&threadid=75222).

The command line doesn't look right to me (should the mplayer be on the next line and piping to audiodump.wav......) but the -channels 6 is what I think you need to add to your AC3 to PCM mplayer line.

Let us know if this works :D

G.
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GordSki
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PostPosted: Tue Jan 18, 2005 10:53 pm    Post subject: Reply with quote

Had a bit of a play my self and I got it working!

Here are the commands I used:

Code:

$ mplayer -vc dummy -vo null -ao pcm -nowaveheader -channels 6 -rawaudio on:format=0x2000 ac3_48k_diatonis_io_anfos.ac3
$ oggenc -q 4 -C 6 -R 48000 audiodump.pcm


Note the '-R' on the oggenc command. You need this to set the sample rate if the ac3 isn't 44.1KHz.

G.
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Kraymer
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PostPosted: Tue Jan 18, 2005 11:05 pm    Post subject: respect Reply with quote

GordSki wrote:
Here are the commands I used: ...

Respect, man! You successfully made your way through the largest manpage I've ever seen. Almost worth a wiki entry? That's quite an alternative when you rip dvds and want to retain 5.1 sound!
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GordSki
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PostPosted: Wed Jan 19, 2005 2:03 pm    Post subject: Re: respect Reply with quote

smash032 wrote:

Respect, man! You successfully made your way through the largest manpage I've ever seen. Almost worth a wiki entry? That's quite an alternative when you rip dvds and want to retain 5.1 sound!


Cheers :D

Just for completeness here are the file sizes I ended up with:


    AC3 - 28 Meg
    OGG - 27 Meg


A bit disappointing, but I'm sure there are savings to be made by dropping the q value.......

The next thing to figure out is how to get 6 channel PCM out of the optical connector on my box so I can listen to all the channels in the OGG 8O

G.
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Kraymer
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PostPosted: Wed Jan 19, 2005 6:51 pm    Post subject: Re: respect Reply with quote

GordSki wrote:
A bit disappointing, but I'm sure there are savings to be made by dropping the q value.......

Sure there is. Now you have every choice oggenc has to offer :)
Quote:
The next thing to figure out is how to get 6 channel PCM out of the optical connector on my box so I can listen to all the channels in the OGG 8O

Yeah, do it! Never stop hacking! ;)

Sebastian
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Sujao
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PostPosted: Wed Jan 19, 2005 11:20 pm    Post subject: Reply with quote

@Gordski: Yeah, I tried your solution too, but I thought it didn't work cause when I play the resulting file I hear an echo and the speech is very quiet. So either the encoding went wrong or the mplayer is not capable of donwmixing the ogg-stream to 2 channels. Do you have a 5.1 audiosystem?

And how did you get the raw ac3 file?

I used:
Code:
mplayer -aid 128 -dumpaudio -dumpfile dumpfile_ac3 VTS_01_5.VOB
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GordSki
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PostPosted: Wed Jan 19, 2005 11:59 pm    Post subject: Reply with quote

I got the ac3 file from a website: http://www.kellyindustries.com/sounds.html. I do have a 5.1 amplifier but I can't anything other than AC3 out of my PC to it. I've played the resulting file through mplayer and it sounded like it down mixed it alright.....

Though with music it can be hard to spot the errors.....

What sort of content (movie, music, etc) is it your trying to encode?

G.

EDIT: I've just had a look at the mplayer man page and I spotted this:

Code:

-channels <number>

[snip]

 MPlayer  asks  the decoder to decode the audio into as many channels as specified.  Now it's up to the decoder to fulfill the requirement.  If the decoder outputs more channels than requested, the exceeding channels are truncated.  This is usually only important when playing videos with AC3  audio (like DVDs).  In that case liba52 does the decoding by default and correctly downmixes the audio into the requested number of channels. NOTE: This option is honored by codecs (AC3 only) filters (surround) and ao drivers (OSS at least).


Looks like mplayer will only be giving you the left and right channels without a downmix. That would explain the quiet voices as they mostly come from the centre channel.......

There might be an option for the OGG decoder to get it to downmix. I'll have a look.
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Sujao
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PostPosted: Thu Jan 20, 2005 12:33 am    Post subject: Reply with quote

It's a movie. I suppose the ogg decoder of mplayer is not able to downmix the 5.1-ogg stream, cause when I play ac3 it downmixes the signals just fine. I have only two speakers. Do you have an idea where to get another ogg-decoder and make mplayer use it. Or first how to verify that its due to the decoder.
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PostPosted: Thu Jan 20, 2005 7:41 am    Post subject: Reply with quote

I think I have tracked the problem now. As the decoder is responsible for the downmixing of the channels (6->2) it must be the decoder for raw pc and ogg used by mplayer. When I play the raw ac3 stream and chose to play only 2 channels everything is normal. Mplayer tells lib52 to decode the video and downmix to 2 channels. Everything is normal. When I do the same with raw pcm the decoder doesnt do that and I hear 2 of 6 channels. The data of the other 4 channels is lost. Same with OGG. The differnce between downmixed 2 channels and cut 2 channels is hearable. The cut 2 channels contain much more envoiromental noise. I suppose I hear the sound of the front speackers.

Here are all the commands I use:

Dump raw AC3 from VOBs:
Code:
mplayer -aid 128 -dumpaudio -dumpfile dumped_ac3 /mnt/cdrom/VIDEO_TS/VTS_01_1.VOB

Play raw AC3
Code:
mplayer -vc dummy -vo null -channels 2 -rawaudio on:format=0x2000 dumped_ac3

convert raw AC3 to raw PCM
Code:
mplayer -vc dummy -vo null -ao pcm -nowaveheader -channels 6 -rawaudio on:format=0x2000 dumped_ac3

play raw PCM
Code:
mplayer -rawaudio on:rate=48000:channels=6  audiodump.pcm

encode to OGG
Code:
oggenc -C 6 -R 48000 -q 4 audiodump.pcm



So the problem is not the missing channels, they are all there but mplayer that has no decoder which could downmix the channels of PCM or OGG correctly! Well what should I do? Any ideas?

Here are the outputs of mplayer when playing:

raw dumped AC3 with -channels 2:
Code:
Playing dumped_ac3.
Cache fill:  7.52% (630784 bytes)    ==========================================================================
Opening audio decoder: [liba52] AC3 decoding with liba52
Using 3DNowEx optimized IMDCT transform
AC3: 5.1 (3f+2r+lfe)  48000 Hz  384.0 kbit/s
Using MMX optimized resampler
AUDIO: 48000 Hz, 2 ch, 16 bit (0x10), ratio: 48000->192000 (384.0 kbit)
Selected audio codec: [a52] afm:liba52 (AC3-liba52)
==========================================================================
Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...
AF_pre: af format: 2 bps, 2 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 2ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...
Video: no video
Starting playback...

raw dumped AC3 with -channels 6:
Code:
Playing dumped_ac3.
==========================================================================
Opening audio decoder: [liba52] AC3 decoding with liba52
Using 3DNowEx optimized IMDCT transform
AC3: 5.1 (3f+2r+lfe)  48000 Hz  384.0 kbit/s
Using MMX optimized resampler
AUDIO: 48000 Hz, 6 ch, 16 bit (0x10), ratio: 48000->576000 (384.0 kbit)
Selected audio codec: [a52] afm:liba52 (AC3-liba52)
==========================================================================
Checking audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/6ch/16bit...
AF_pre: af format: 2 bps, 6 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 6ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 6ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/6ch/16bit...
Video: no video
Starting playback...

raw PCM with -channels 2:
Code:
Playing audiodump.pcm.
Cache fill:  0.88% (73728 bytes)    ==========================================================================
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 48000 Hz, 6 ch, 16 bit (0x10), ratio: 576000->576000 (4608.0 kbit)
Selected audio codec: [pcm] afm:pcm (Uncompressed PCM)
==========================================================================
Checking audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/2ch/16bit...
AF_pre: af format: 2 bps, 6 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 6ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/2ch/16bit...
Video: no video
Starting playback...

raw PCM with -channels 6:
Code:
Playing audiodump.pcm.
Cache fill: 10.45% (876544 bytes)    ==========================================================================
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 48000 Hz, 6 ch, 16 bit (0x10), ratio: 576000->576000 (4608.0 kbit)
Selected audio codec: [pcm] afm:pcm (Uncompressed PCM)
==========================================================================
Checking audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/6ch/16bit...
AF_pre: af format: 2 bps, 6 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 6ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 6ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/6ch/16bit...
Video: no video
Starting playback...

OGG with -channels 2:
Code:
Playing audiodump.ogg.
Cache fill:  6.15% (516096 bytes)    Ogg file format detected.
==========================================================================
Opening audio decoder: [libvorbis] Ogg/Vorbis audio decoder
AUDIO: 48000 Hz, 6 ch, 16 bit (0x10), ratio: 64500->576000 (516.0 kbit)
Selected audio codec: [vorbis] afm:libvorbis (OggVorbis Audio Decoder)
==========================================================================
Checking audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/2ch/16bit...
AF_pre: af format: 2 bps, 6 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 6ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/2ch/16bit...
Video: no video
Starting playback...

OGG with -channels 6:
Code:
Playing audiodump.ogg.
Cache fill: 10.16% (851968 bytes)    Ogg file format detected.
==========================================================================
Opening audio decoder: [libvorbis] Ogg/Vorbis audio decoder
AUDIO: 48000 Hz, 6 ch, 16 bit (0x10), ratio: 64500->576000 (516.0 kbit)
Selected audio codec: [vorbis] afm:libvorbis (OggVorbis Audio Decoder)
==========================================================================
Checking audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/6ch/16bit...
AF_pre: af format: 2 bps, 6 ch, 48000 hz, little endian signed int
AF_pre: 48000Hz 6ch Signed 16-bit (Little-Endian)
AO: [oss] 48000Hz 6ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 48000Hz/6ch/16bit -> 48000Hz/6ch/16bit...
Video: no video
Starting playback...


The important difference is at the following line:
Code:
Checking audio filter chain for 48000Hz/2ch/16bit -> 48000Hz/2ch/16bit...

With AC3 mplayer gets the signal already with 2ch, because the decoder downmixes it. With PCM and OGG mplayer gets the signal with 6 channels, tries to downmix it but fails.
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GordSki
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PostPosted: Thu Jan 20, 2005 10:18 am    Post subject: Reply with quote

I managed to find a vob file with an ac3 test tone (ie a hiss and a voice saying which speaker the his is coming from) and ran that through the ogg encode process.

When I played the ogg file with no options, mplayer cut all but the left and right channels. With the '-channels 6' I still only got the left and right, however I think mplayer was giving all the channels to alsa but alsa was dropping the ones it couldn't play. I guess if you have a 5.1 capable (and configured) sound card you would get all the channels.

I think what needs to be done now is to figure out if there is a way to get a downmix filter into the audio chain.

More fun :cry:

G.

PS. I got the test file here: http://www.digital-digest.com/dvd/downloads/trailers.html
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PostPosted: Thu Jan 20, 2005 10:29 am    Post subject: Reply with quote

Vorbis 5.1 is of no practical use currently. In contrast to AC3 it offers no channel coupling (not even the 2 channel coupling used in stereo mode!) so good space savings while preserving quality is *not* possible. Monty wanted to do some 5.1 work into Vorbis, but till now nothing happened.
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PostPosted: Thu Jan 20, 2005 10:48 am    Post subject: Reply with quote

Yet again 'man mplayer' might have the answer:

Code:

-af <filter1[=option1:option2:...],filter2,...>
 Activate a comma separated list of audio filters and their options.
 Available filters are:

[snip]

          channels[=nch]
               Change  the number of channels to nch output channels.
           If the number of output channels is bigger than the number
           of input channels empty channels are inserted  (except mixing
           from  mono to stereo, then the mono channel is repeated in both
           of the output channels).  If the number of output channels is
           smaller  than  the  number  of  input channels the exceeding
           channels are truncated.

[snip]

          pan[=n:l01:l02:..l10:l11:l12:...ln0:ln1:ln2:...]
               Mixes channels arbitrarily, see DOCS/HTML/en/devices.html#audio-dev for details.
                  n: number of output channels (1-6).
                  lij: how much of input channel j is mixed into output channel i.


I'll have to wait till I get home to give it a shot tho.....

G.
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Sujao
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PostPosted: Thu Jan 20, 2005 10:52 am    Post subject: Reply with quote

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In contrast to AC3 it offers no channel coupling (not even the 2 channel coupling used in stereo mode!)


What is coupling? Is that a synonym for downmixing?

I think I will leave AC3 in my movie. Firstly it is too much work and I am tired of it. Secondly I dont seem to gain much space by it anyway.
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PrakashP
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PostPosted: Thu Jan 20, 2005 10:56 am    Post subject: Reply with quote

coupling is a common technique used to reduce data. Basically some frequence bands of channels which contain more or less the same data, will be encoded as one band instead of individual bands. That's why ac3 isn't really discrete, but on the other hand offers great quality/bitrate.
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Sujao
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PostPosted: Thu Jan 20, 2005 11:01 am    Post subject: Reply with quote

Ah I heard of that but didnt know thats the name for it. :)
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Kraymer
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PostPosted: Thu Jan 20, 2005 11:08 am    Post subject: joint stereo? Reply with quote

From what you describe.. is it the same as 'Joint stereo'?
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