On 2014-04-24 12:33:43 +0200, Niels Möller wrote: > Janne Grunau <[email protected]> writes: > > > the raw audio muxer have separate names for the sample formats. > > > > -f f32le $FILE at the end of the command will output little endian float > > samples. > > Nice, -f s24le seems to work too. But if I also want a proper .au (or > .wav) header, I'll have to add that myself (e.g, using sox should work, > I guess)?
for wav or au you have to override the audio codec to pcm_f32le, i.e. -i $INFILE -c:a pcm_f32le OUT.wav See avconv -codecs | grep ' pcm_' for all sample format specific codecs. > > Or you could add an upsample to 96kHz filter, avconv should be smart > > enough not to use it when the sampling rate changes to 96kHz. > > Is that -filter 'aformat=sample_rates=96000' ? Gives me an interesting > error, though... yes, works here as expected with a random non dts / non-96kHz sample > $ ./avconv -i Master\ Audio\ 5.0\ 96khz.dts -y -filter > 'aformat=sample_rates=96000' out.wav > avconv version v10_alpha2-146-g274f5c9, Copyright (c) 2000-2014 the Libav > developers > built on Apr 24 2014 10:16:35 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9) > [dca @ 0x23aa0c0] DCA XLL: number of xll samples (1024) != number of core > samples (512) > resampling required. > [dts @ 0x2399080] max_analyze_duration reached > [dts @ 0x2399080] Estimating duration from bitrate, this may be inaccurate > Input #0, dts, from '/data/nisse/libav-dev-shared/Master Audio 5.0 > 96khz.dts': > Duration: 00:00:27.30, start: 0.000000, bitrate: 1535 kb/s > Stream #0.0: Audio: dca (DTS-HD MA), 48000 Hz, 5.0, fltp, 1536 kb/s > Output #0, wav, to 'out.wav': > Metadata: > ISFT : Lavf55.12.0 > Stream #0.0: Audio: pcm_s16le, 96000 Hz, 5.0, s16, 7680 kb/s > Stream mapping: > Stream #0:0 -> #0:0 (dca -> pcm_s16le) > Press ctrl-c to stop encoding > [dca @ 0x23aa0c0] DCA XLL: number of xll samples (1024) != number of core > samples (512) > resampling required. > Input stream #0:0 frame changed from rate:48000 fmt:fltp ch:5 chl:5.0 to > rate:96000 fmt:fltp ch:5 chl:5.0 > *** Error in `./avconv': corrupted double-linked list: 0x00000000023ad2e0 > *** > ^C interesting, can you check if that happens with -ar 96000 too? a run with '-v verbose' might be interesting too. I guess the audio filter can't handle changes and avconv doesn't reconfigures it on sample rate changes > BTW, what would it take to enable wav output in other sample formats, > configured via -sample_fmt or -filter aformat=sample_fmts=...? I notice > that ff_codec_wav_tags (definied in libavformat/riff.c) includes a tag > for, e.g, AV_CODEC_ID_PCM_S24LE, but I don't understand how it fits > together. see above Janne _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
