I am capturing a live audio stream to Opus, and no matter what I choose for the audio sample rate, I get 48khz in the output file.
This is my command line ./ffmpeg -f alsa -ar 16000 -i sysdefault:CARD=CODEC -f alsa -ar 16000 -i sysdefault:CARD=CODEC_1 -filter_complex join=inputs=2:channel_layout=stereo:map=0.1-FR\|1.0-FL,asetpts=expr=N/SR/TB -ar 16000 -ab 64k -c:a opus -vbr off -compression_level 5 output.ogg And this is what ffmpeg responds with: Output #0, ogg, to 'output.ogg': Metadata: encoder : Lavf57.48.100 Stream #0:0: Audio: opus (libopus), 16000 Hz, stereo, s16, delay 104, padding 0, 64 kb/s (default) Metadata: encoder : Lavc57.54.100 libopus However, it appears that ffmpeg has lied, because when analysing the file again, I get: Input #0, ogg, from 'output.ogg': Duration: 00:00:03.21, start: 0.000000, bitrate: 89 kb/s Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, delay 156, padding 0 Metadata: ENCODER : Lavc57.54.100 libopus I should also mention, that VLC also reports the file as being 48Khz - and the file size is way too big to be 16khz - so I'm pretty confident that it actually contains 48000 samples per second of data.... I decoded the file to WAV, then used opusenc to compress it back to 16Khz, and it was way smaller.... I have tried so many permutations of sample rate, simplifying down to a single audio input etc etc - always with the same result. Thanks, _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
