On Apr 20, 2013, at 1:01 PM, Brad O'Hearne <[email protected]> wrote:
> 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. In lieu of making no progress, and being pretty much out of options, and nothing being apparent to anyone else either, perhaps I can retrench with some simpler questions in hopes of moving forward: 1. Is anyone aware of audio being captured from QuickTime, encoded successfully to AAC in an FLV file? I wouldn't think this is the case, but maybe there's some prohibiting factor that prevents the encoder from being able to accomplish this. 2. How do you get rid of distortion / buzz in encoded AAC audio? I'm at a loss -- I've taken working code, changed the encoder / sample format / sample rate and gone from perfect audio to distorted audio -- same input, same processing code. 3. Is there any way to determine from the recorded/written FLV file what the reason for distortion / buzz is? I have no experience analyzing the end product of encoding -- is there a way to pinpoint what is happening in encoding that creates this distortion where before it didn't exist? Thanks, Brad _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
