libav

I am decoding aac to pcm with ffmpeg with avcodec_decode_audio3. However it decodes into AV_SAMPLE_FMT_FLTP sample format (PCM 32bit Float Planar) and i need AV_SAMPLE_FMT_S16 (PCM 16 bit signed - S16LE). I know that ffmpeg can do this easily with -sample_fmt. I want to do the same with the code but i still couldn't figure it out. audio_resample did not work for: it fails with error message: .... conversion failed. EDIT 9th April 2013 : Worked out how to use libswresample to do this... much faster! At some point in the last 2-3 years FFmpeg's AAC decoder's output format changed from AV_SAMPLE

I changed in my code avcodec_decode_audio3 to avcodec_decode_audio4 and added the frame handling. But now I cannot decode AAC frames anymore. Why does avcodec_decode_audio4 return -22 ( invalid argument )? Following the answer below, does this have something to do with the parameters in AVContext that need to be set? I had to use avcodec_decode_audio4 because I updated my ffmpeg and then got the following error: [NULL @ 0xb14f020] Custom get_buffer() for use withavcodec_decode_audio3() detected. Overriding with avcodec_default_get_buffer [NULL @ 0xb14f020] Please port your application to

I have downloaded videos from CDN in flv format (video H264 and audio AAC) and remux to them to MP4 format. But videos are limited by length. So i've downloaded each video in several parts: started at start point, at point 1, at point 2 (by using seek parameter in url). Each point starts little earlier than ending of previous one. Using av_read_frame i scanned all parts and found that intersecting packets not only have same sizes and order but also their dts/pts shifted from each other by constant value. So to concat starting video with video started at point 1 I must do following: 1. Create

By default, libavformat writes error messages to stderr , Like: Estimating duration from bitrate, this may be inaccurate How can I turn it off? or better yet, pipe it to my own neat logging function? Edit: Redirecting stderr to somewhere else is not acceptable since I need it for other logging purposes, I just want libavformat to not write to it. Looking through the code, it appears you can change the behavior by writing your own callback function for the av_log function. From the description of this function in libavutil/log.h : Send the specified message to the log if the level is less than

Note : I'm aware ffmpeg and libav are different libraries . This is a problem common to both. Disclaimer : Duplicate of SO question marked as answered but actually didn't give a proper solution. Insufficient UDP buffer size causes broken streams for several high resolution video streams. In LibAV/FFMPEG it's possible to set the udp buffer size for udp urls (udp://...) by appending some options (buffer_size) to it. However, for RTSP urls this is not supported . These are the only solutions I've found: Rebuilding ffmpeg/libav changing the UDP_MAX_PKT_SIZE in the udp.c source file. Using a nasty