On 2014-04-24 11:13:23 +0200, Niels Möller wrote: > When testing the dca code, I would really like to generate an > uncompressed file (.wav or .au) with more than 16 bits per sample, > ideally 24 bits signed integers. > > avconv -sample_fmts lists s32 and flt, and flt is what dcadec.c actually > generates, so that sounds like a natural choice. However, after testing > some different placements of -sample_fmt flt on the command line, I > still can't get float output; it either claims the format is not > supported, or accepts the option but generates s16 samples nonetheless. > > So how am I supposed to do this? Do I need to specify some other filter > on the command line as well?
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. > In addition, it would be nice if I could force the output sample rate to > 96kHz, disabling the downsampling to 48 kHz. Is that possible? (Probing > thinks the file is 48kHz, but the decoder actually generates 96kHz, and > avconv is then friendly enough to downsample it for me). I fear the easiest way to fix that is forcing 96kHz during probing. avconv doesn't support changing stream parameters on the fly. 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. HTH Janne _______________________________________________ libav-devel mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-devel
