Hi to all, I have a problem that I can not solve. In my encoder I'm encoding video and audio, the problem comes with the audio part. My original audio raw samples are pcm float -1/+1 stereo. So I have 4 bytes (2 for the L channel and 2 for the R channel TOTAL = 4 bytes). To encode an audio frame and use the avcodec_encode_audio ffmpeg need the samples to be signed short (2 bytes - 1 for the L channel and 1 for the R channel TOTAL = 2 bytes) so to use this routine I need to convert my original raw samples from float to signed short. I use this simple algorithm
sample_signed_short = (signed short) sample_float * 32767; // this for every sample I have Doing this I'm sure I loose lots of quality because 1 byte can not store the information that 2 bytes contains right? In fact the quality of the audio of my produced mpeg is very bad. What can I do? Where are my mistakes? Please I need some help. Regards, Franco _______________________________________________ libav-user mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/libav-user
