Hi guys, I'm using LAME (standalone executable) to do audio encoding and use ffmpeg for muxing. Until recently I noticed that LAME introduce some encoder delay inside the audio (so that some silence inserted at the beginning of the audio frame)
My program also uses ffmpeg API to open file and read frames/samples. Seems by default ffmpeg doesn't handle this so that the silence got returned when samples are decoded. So I'd like to know, how to instruct ffmpeg to ignore this delay and only return real data to me? If different encoders use different delays, can ffmpeg handle them automatically? Or when muxing using ffmpeg, I should set the encoder delay myself? I saw in mp3dec there're code to get encoder delay values. But I'm writing to an AVI file so I guess decoding is done by mpegaudiodec_template, inside which seems there's nothing to handle the delay. Thanks
_______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
