I am trying to use ffmpeg, and have been doing a lot of experiment last 1 month. I have not been able to get through. Is it really difficult to use FFmpeg?
My requirement is simple as below. Can you please guide me if ffmpeg is suitable one or I have implement on my own (using codec libs available). 1. I have a webm file (having VP8 and OPUS frames) 2. I will read the encoded data and send it to remote guy 3. The remote guy will read the encoded data from socket 4. The remote guy will write it to a file (can we avoid decoding). 5. Then remote guy should be able to pay the file using ffplay or any player. Now I will take a specific example. 1. Say I have a file small.webm, containing VP8 and OPUS frames. 2. I am reading only audio frames (OPUS) using av_read_frame api (Then checks stream index and filters audio frames only) 3. So now I have data buffer (encoded) as packet.data and encoded data buffer size as packet.size (Please correct me if wrong) 4. Here is my first doubt, everytime audio packet size is not same, why the difference. Sometimes packet size is as low as 54 bytes and sometimes it is 420 bytes. For OPUS will frame size vary from time to time? 5. Next say somehow extract a single frame (really do not know how to extract a single frame) from packet and send it to remote guy. 6. Now remote guy need to write the buffer to a file. To write the file we can use av_interleaved_write_frame or av_write_frame api. Both of them takes AVPacket as argument. Now I can have a AVPacket, set its data and size member. Then I can call av_write_frame api. But that does not work. Reason may be one should set other members in packet like ts, dts, pts etc. But I do not have such informations to set. Can somebody help me to learn if FFmpeg is the right choice, or should I write a custom logic like parse a opus file and get frame by frame. _______________________________________________ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
