title: ffmpeg audio conversion distorted - half rate
body: <p>I'm trying to convert an asf audio to mp3 using ffmpeg. 
  But I have one specific audio that gets distorted in the middle and starts like if the person was talking in slow motion (at half rate). </p>
  
  <p>The command I'm using is:</p>
  
  <p>ffmpeg - i input.asf -ac 2 output.mp3</p>
  
  <p>I've tried a lot of options, but about the middle of the audio is when it fails. 
  The raw file sounds good, so it's not the recording. It is af in the middle of the transmission the frame rate went down for some reason. </p>
  
  <p>Thanks all!</p>
  
  <p>[EDIT]</p>
  
  <p>I'm adding the console response after running the command as a suggestion from LordNeckbeard:</p>
  
  <pre><code>[root@mynasserver home]# ffmpeg -i recording-8532-1.asf -ac 2 -ab 64k -ar 44100 recording-8532-ac2-ar44100.mp3
  FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
  built on Jan 29 2012 23:56:18 with gcc 4.1.2 20080704 (Red Hat 4.1.2-51)
  configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-avfilter-lavf --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
  libavutil     50.15. 1 / 50.15. 1
  libavcodec    52.72. 2 / 52.72. 2
  libavformat   52.64. 2 / 52.64. 2
  libavdevice   52. 2. 0 / 52. 2. 0
  libavfilter    1.19. 0 /  1.19. 0
  libswscale     0.11. 0 /  0.11. 0
  libpostproc   51. 2. 0 / 51. 2. 0
  [flv @ 0x86a4850]max_analyze_duration reached
  [flv @ 0x86a4850]Estimating duration from bitrate, this may be inaccurate
  Input #0, flv, from 'recording-8532-1.asf':
  Metadata:
  source          : STW MediaProxy v3.3.7.19894
  Duration: 04:00:08.49, start: 0.000000, bitrate: N/A
  Stream #0.0: Audio: aac, 44100 Hz, 2 channels (FC), s16
  Output #0, mp3, to 'recording-8532-ac2-ar44100.mp3':
  Metadata:
  TSSE            : Lavf52.64.2
  Stream #0.0: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 64 kb/s
  Stream mapping:
  Stream #0.0 -&gt; #0.0
  Press [q] to stop encoding
  size=  150906kB time=19315.93 bitrate=  64.0kbits/s    
  video:0kB audio:150906kB global headers:0kB muxing overhead 0.000021%
  </code></pre>
  
  <p>So from the data above, you can see the input file is about 4hrs. The output ends up being around 5 hrs 20 mins.</p>