Hi, We are using libav to encode / decode opus audio between two SIP clients.
The encoding client use libopus while the decoding client use either libopus (before libavcodec56) or opus native decoder (since libavcodec56). Audio encoding format is signed 16 bits (AV_SAMPLE_FMT_S16), sample rate = 48000 Using libopus for decoding, everything work fine. But using native libav opus decoding, we see lot of noise in decoded audio. I compared encoder and decoder parameters set in libav. I saw first that FMT used for decoding is AV_SAMPLE_FMT_FLTP (defined in opus_decode_init). Replacing this value by AV_SAMPLE_FMT_S16 is not successful. Otherwise all other parameters look correct. Does opus native decoder support AV_SAMPLE_FMT_S16 ? Any advice to investigate this issue ? Thanks, Eloi _______________________________________________ libav-api mailing list [email protected] https://lists.libav.org/mailman/listinfo/libav-api
