On Fri Apr 15 bat guano wrote: >> Simon wrote: > > > rtmpdump produces FLV files, which have nowhere to put a value for the > average bit rate. So this isn't something that rtmpdump could fix. > > I think ffmpeg is the right place to handle this. When ffmpeg runs > (using -acodec copy), it displays a status line of the form > size= 33270kB time=821.71 bitrate= 331.7kbits/s > where these values are updated every second or so. So ffmpeg does > know the average bit rate, but it doesn't store it in the m4a file. >
> I also don't think it's a problem caused by streamed files from > RTMPDump. > > And it's not just '-acodec copy' that gives no avgBitrate. > FFmpeg doesn't seem to set the avgBitrate when converting other files > to m4a. > Using commands like this:- > ffmpeg -i foo -acodec libfaac foo.m4a OK. I'll eat my words. I forgot about that. I've done some more testing of a couple of mp3 files (not downloaded with get_iplayer) using that command: ffmpeg -i 128bit.mp3 -acodec libfaac 128bit.m4a ffmpeg -i 128kbs.mp3 -acodec libfaac 128kbs.m4a Both m4a files would not play in the Marantz, and the decConfigDescr in the esds atom shows the same problem - avgBitrate = 0 bufferSizeDB = 0 (0x000000) <24 bits> maxBitrate = 64000 (0x0000fa00) avgBitrate = 0 (0x00000000) _______________________________________________ get_iplayer mailing list [email protected] http://lists.infradead.org/mailman/listinfo/get_iplayer
