On Apr 20, 2013, at 9:36 AM, Brad O'Hearne <[email protected]> wrote:
> So after these changes, the audio is distorted. I'm not sure why -- as I know > the actual capture-resample-encode-write approach / pipeline is sound, my > guess is that this is some kind of additional configuration or handling need > surrounding AAC. I was able to eek just a bit closer to the goal by changing the AAC codec context's bit rate from 192 to 128 and the sample rate from 44100 to 9600. I found these values from looking at the defaults on the codec itself when it was loaded, so I inferred that the codec context would appreciate such values. At least now the audio timing sounds right (it isn't slower like it was before). However, the audio still suffers from noise -- a buzzing over the expected audio, still like someone talking through a running fan. I have uploaded this FLV file (which contains only audio) so that anyone interested can listen. https://github.com/BigHillSoftware/QTFFmpeg The output FLV can be found in the code above at: Sample Output/Output.flv I'm not sure what to try next. While I mentioned above that I had changed the bit rate, that didn't really seem to be a big factor -- the change seemed to be most affected by the sample rate changes (I experimented with various values, and that seemed to have the greatest effect). Note that the source audio sample rate was 44100, so I'm not sure why using a sample rate of 44100 on the codec context was problematic. Beyond this, I'm pretty much grabbing at straws. I'm suspecting that the answer is probably just a knob or switch somewhere, a setting that will make it work. I read a bit about bit_rate_tolerance, but the source code doc doesn't say anything about what values to use, so I didn't know how to set it. Any ideas would be greatly appreciated. Thanks, Brad _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
