On Sat, 17 Jan 2009 18:54:54 -0500, Franco Amato <[email protected]> wrote: > 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)
A byte is 8 bits. A short is 16 bits. A float is 32 bits. ffmpeg uses "16 bit sound", where each sample is a signed integer between -0x8000 and 0x7FFF. There is one sample for each channel. If you have 2 16-bit channels at 22050 hz, then you will have 88200 bytes per second for stereo sound. -- Michael Conrad IntelliTree Solutions llc. 513-552-6362 [email protected] _______________________________________________ libav-user mailing list [email protected] https://lists.mplayerhq.hu/mailman/listinfo/libav-user
