FFmpeg/Libav audio decode example

Question

I am trying to get the same output as that of ffmpeg/avconv from converting an MP2 file to raw PCM using code. I used the audio_decode_example function from the tutorial included in the FFmpeg/Libav documentation but the outputs were different.

What steps does the ffmpeg/avconv command perform that I need to add to my program to get the same output?

Edit: This is the output from avconv -v debug -i test.mp2 test.sw:

avconv version v10_beta1-302-g606cc8a, Copyright (c) 2000-2014 the Libav developers
built on Mar 20 2014 20:28:32 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
configuration: 
libavutil     53.  6. 0 / 53.  6. 0
libavcodec    55. 34. 1 / 55. 34. 1
libavformat   55. 12. 0 / 55. 12. 0
libavdevice   54.  0. 0 / 54.  0. 0
libavfilter    4.  3. 0 /  4.  3. 0
libavresample  1.  2. 0 /  1.  2. 0
libswscale     2.  1. 2 /  2.  1. 2
Splitting the commandline.
Reading option '-v' ... matched as option 'v' (set libav* logging level) with argument 'debug'.
Reading option '-i' ... matched as input file with argument 'test.mp2'.
Reading option 'test1.sw' ... matched as output file.                                  
Finished splitting the commandline.                                       
Parsing a group of options: global .                
Applying option v (set libav* logging level) with argument debug.
Successfully parsed a group of options.
Parsing a group of options: input file test.mp2.
Successfully parsed a group of options.
Opening an input file: test.mp2.
[mp3 @ 0x27e5060] Probed with size=2048 and score=51
[mp3 @ 0x27e5060] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, mp3, from 'test.mp2':
  Duration: 00:00:04.80, start: 0.000000, bitrate: 64 kb/s
    Stream #0.0, 200, 1/14112000: Audio: mp2, 48000 Hz, stereo, s16p, 64 kb/s
Successfully opened the file.
Parsing a group of options: output file test1.sw.
Successfully parsed a group of options.
Opening an output file: test1.sw.
Successfully opened the file.
Detected 2 logical cores.
[abuffer @ 0x27e5860] tb:1/48000 samplefmt:s16p samplerate: 48000 ch layout:0x3
[abuffersink @ 0x2807040] auto-inserting filter 'auto-inserted fifo 0' between the filter 'audio format for output stream 0:0' and the filter 'output stream 0:0'
[aformat @ 0x2806bc0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed filter 0 anull' and the filter 'audio format for output stream 0:0'
[AVAudioResampleContext @ 0x28092a0] audio_convert: found function: s16p to s16  (C)
[AVAudioResampleContext @ 0x28092a0] audio_convert: found function: s16p to s16  (SSE2)
[resample @ 0x2808560] fmt:s16p srate:48000 cl:stereo -> fmt:s16 srate:48000 cl:stereo
Output #0, s16le, to 'test1.sw':
  Metadata:
    encoder         : Lavf55.12.0
    Stream #0.0, 0, 1/90000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
  Stream mapping:
    Stream #0:0 -> #0:0 (mp2 -> pcm_s16le)
Press ctrl-c to stop encoding
Input stream #0:0 frame changed from rate:48000 fmt:s16p ch:2 chl:stereo to rate:48000 fmt:s16 ch:2 chl:stereo
Detected 2 logical cores.
[abuffer @ 0x2806e40] tb:1/48000 samplefmt:s16 samplerate: 48000 ch layout:0x3
[abuffersink @ 0x2806fa0] auto-inserting filter 'auto-inserted fifo 0' between the filter 'audio format for output stream 0:0' and the filter 'output stream 0:0'
No more output streams to write to, finishing.
size=     900kB time=4.80 bitrate=1536.0kbits/s    
video:0kB audio:0kB global headers:0kB muxing overhead 0.000000%

Answer 1

I had the same problems. It seems libav mp2 decoding by default outputs in S16P format (planar) which is not so common in various audio editing applications that support raw input (audacity for example). I solved my problem by specifiying request_sample_fmt in AVCodecContext being AV_SAMPLE_FMT_S16. Thus there's no additional need for software resampling.

In function audio_decode_example:

...
c = avcodec_alloc_context3(codec);
if (!c) {
    fprintf(stderr, "Could not allocate audio codec context\n");
    exit(1);
}
c->request_sample_fmt = AV_SAMPLE_FMT_S16;
...