[Libav-user] Audio quality loss while encoding
klaussfreire at gmail.com
Mon Apr 29 18:44:13 CEST 2013
On Thu, Apr 25, 2013 at 2:16 PM, Paul B Mahol <onemda at gmail.com> wrote:
> On 4/25/13, Claudio Freire <klaussfreire at gmail.com> wrote:
>> On Thu, Apr 25, 2013 at 6:29 AM, Paul B Mahol <onemda at gmail.com> wrote:
>>> I listened the sample from mentioned github repo. And its evident that
>>> there are
>>> either holes (end of every? channel data is cut off) or extra noise
>>> after each channel is added.
>>> Because this does not happen with any of ffmpeg libraries or tools I can
>>> conclude with 1000% confidence that bug is in your code.
>> It does happen to me with "ffmpeg" (the tool - no code of mine), when
>> encoding to AAC. And the symptom is very similar to that sample
> Plese open bug report, with exact step to reproduce bug.
Revisiting this issue, it doesn't sound at all like the previously
linked example output. I was confused, it's similar, but seems to be
related to quantization rather than buffer misalignment. It sounds as
if some components were allocated too few bits, in spite of a
relatively high bitrate selection (256k).
I will post a minimal test case in a few days (I've been busy with RL issues).
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