On 12/07/2010 13:30, Mark Kenna wrote:
Hi
I'm using a recent build of FFMpeg to try and convert PCM to ACC. The
problem I am having is that the encoding will "stall" part way into
the encoding.
I have confirmed that this is happening through FFMpeg.exe as well as
avcodec_encode_audio(...). Here is a copy of what FFMPEG.exe is showing:
[wav @ 0049d4b0]max_analyze_duration reached
[wav @ 0049d4b0]Estimating duration from bitrate, this may be inaccurate
Input #0, wav, from '814944.5298.wav':
Duration: 00:03:11.00, bitrate: 88 kb/s
Stream #0.0: Audio: pcm_u8, 11025 Hz, 1 channels, u8, 88 kb/s
Output #0, adts, to 'test.aac':
Metadata:
encoder : Lavf52.67.0
Stream #0.0: Audio: aac, 11025 Hz, 1 channels, s16, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
^C
If I attempt to do the conversion using avcodec, I get around 60kb
worth of encoded data (examining the contents of the output file I can
see Lavc in the file header), further calls will result in an
identical 10 bytes worth of data being returned each time.
Does someone have any ideas?
Thanks,
Mark.
I have done some more testing and found that the command line encoder
will not work with MONO audio - only stereo, also certain bit rates will
exhibit the "freezing" behaviour.
I have tried to set the wave format to a higher bit-rate with stereo
sound but still nothing through LibAV. Can someone possibly try this as
it could be a problem with FFmpeg - I can send a bin capture of my mic
output at any time.
Cheers,
Mark.
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