audio encoding with ffmpeg | sox | ffmpeg

cameris · 14th October 2009, 15:35

Hi,

I have a problem to encode audio tracks in my encoding script. It uses ffmpeg to encode to aac and for 5.1 tracks also sox to get the channel order right. I have tested it on a couple of tv-series (longest was 45 min) and everything has worked great.
But now I have tested it on some full featured movies and on every 5.1 audio track it hangs aroung 80MB (output file, bitrate: 128k) and I dont have a clue why. It doesnt give a error message neither quits, it just stops doing anything.

The command for 5.1 looks like this:

Code:

ffmpeg -i $input -async 1 -vsync 1 -v 0 -map $map -vn -sn -ac 6 -acodec pcm_s16le -ar $afreq -f wav - | \
sox -t raw -s -2 -c 6 -r $afreq - -t wav - remix -m 6 5 1 2 3 4 | \
ffmpeg -i - -ac 6 -v 0 -y -acodec libfaac -ab $audioquality $audiofile

Has anybody has an idea why?

nm · 14th October 2009, 15:47

Does the same thing happen if you redirect WAV data from the first ffmpeg process or sox to a file? If not, what happens when you encode that file?

However, I'd recommend switching to neroAacEnc if you're on x86(-64). It's much better than FAAC.

cameris · 14th October 2009, 19:04

I have testet ffmpeg | sox -> wav, no error. But when encoding the wav to aac I've got

Code:

[pcm_s16le @ 0x9970d60]invalid PCM packet.3kbits/s
Error while decoding stream #0.0

I also tried a 6ch -> 2ch downmix without any problem.

Is nero really that much better? I thought the difference is only noticeable on bitrates below 128k.
If I try it, do I still need sox for channel ordering or does nero do the work?

nm · 14th October 2009, 19:53

Quote:

Originally Posted by cameris

I have testet ffmpeg | sox -> wav, no error. But when encoding the wav to aac I've got

Code:

[pcm_s16le @ 0x9970d60]invalid PCM packet.3kbits/s
Error while decoding stream #0.0

Ok, I guess you're hitting the 4 GB WAV (RIFF) file size limit. I'm not sure if there's a workaround other than using raw PCM audio. NeroAACEnc has -ignorelength parameter that should allow you to read larger than 4 GB WAV files.

Quote:

Is nero really that much better? I thought the difference is only noticeable on bitrates below 128k.

Perhaps for stereo, but you're encoding 5.1-channel audio. 128 kbps is actually a pretty low bitrate for that -- especially when using FAAC.

Also, while I haven't done a listening test myself, I'd guess that FAAC loses badly to both Lame and neroAACEnc at 128kbps stereo.

Quote:

If I try it, do I still need sox for channel ordering or does nero do the work?

Nope, you'll probably need to order them first.

cameris · 15th October 2009, 13:26

Quote:

Originally Posted by nm

Ok, I guess you're hitting the 4 GB WAV (RIFF) file size limit. I'm not sure if there's a workaround other than using raw PCM audio. NeroAACEnc has -ignorelength parameter that should allow you to read larger than 4 GB WAV files.

The WAV files had 3.8 GB, so I tried nero and it worked like a charm, only order had to be changed.

eventually my script is working

. Many

...

nm · 15th October 2009, 13:49

Quote:

Originally Posted by cameris

The WAV files had 3.8 GB

Ok, then there may be some lower limit between 2 GB and 4 GB in ffmpeg WAV demuxer.

Quote:

eventually my script is working

. Many

...

Cheers!

Selur · 14th October 2009, 20:43

did you also try to use named pipes instead of | ?

14th October 2009, 15:35	#1 \| Link
cameris Registered User Join Date: Oct 2009 Posts: 3	audio encoding with ffmpeg \| sox \| ffmpeg Hi, I have a problem to encode audio tracks in my encoding script. It uses ffmpeg to encode to aac and for 5.1 tracks also sox to get the channel order right. I have tested it on a couple of tv-series (longest was 45 min) and everything has worked great. But now I have tested it on some full featured movies and on every 5.1 audio track it hangs aroung 80MB (output file, bitrate: 128k) and I dont have a clue why. It doesnt give a error message neither quits, it just stops doing anything. The command for 5.1 looks like this: Code: ffmpeg -i $input -async 1 -vsync 1 -v 0 -map $map -vn -sn -ac 6 -acodec pcm_s16le -ar $afreq -f wav - \| \ sox -t raw -s -2 -c 6 -r $afreq - -t wav - remix -m 6 5 1 2 3 4 \| \ ffmpeg -i - -ac 6 -v 0 -y -acodec libfaac -ab $audioquality $audiofile Has anybody has an idea why?

14th October 2009, 19:04	#3 \| Link
cameris Registered User Join Date: Oct 2009 Posts: 3	I have testet ffmpeg \| sox -> wav, no error. But when encoding the wav to aac I've got Code: [pcm_s16le @ 0x9970d60]invalid PCM packet.3kbits/s Error while decoding stream #0.0 I also tried a 6ch -> 2ch downmix without any problem. Is nero really that much better? I thought the difference is only noticeable on bitrates below 128k. If I try it, do I still need sox for channel ordering or does nero do the work? Last edited by cameris; 14th October 2009 at 19:09.

14th October 2009, 20:43	#7 \| Link
Selur Registered User Join Date: Oct 2001 Location: Germany Posts: 7,277	did you also try to use named pipes instead of \| ? __________________ Hybrid here in the forum, homepage

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14th October 2009, 15:47	#2 \| Link
nm Registered User Join Date: Mar 2005 Location: Finland Posts: 2,641	Does the same thing happen if you redirect WAV data from the first ffmpeg process or sox to a file? If not, what happens when you encode that file? However, I'd recommend switching to neroAacEnc if you're on x86(-64). It's much better than FAAC.