[Libav-user] AAC with FLV

Kalileo kalileo at universalx.net
Mon Apr 22 04:34:06 CEST 2013

On Apr 22, 2013, at 06:32 , Brad O'Hearne wrote:

> On Apr 20, 2013, at 1:01 PM, Brad O'Hearne <brado at bighillsoftware.com> 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. 

I strongly doubt that there is any reason why that should not be possible.

> 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. 

You took working code for a different situation. Instead of playing around with values, I would systematically analyze every step of your process, verify every value, to check if your assumptions are correct.

Memory alignment might be an issue too.

Again, I would check the process every step beginning with the first step, the captured audio. Assuming the source is clean, and your resampling logic is correct, somewhere you feed the process with incorrect values. Verify every single value.

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