[Libav-user] Audio quality loss while encoding

[Paul B Mahol]

On 4/29/13, Claudio Freire <klaussfreire at gmail.com> wrote:
> 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
>>> output.
>>
>> 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).

Note that native aac encoder is experimental and mostly useless.

>
> I will post a minimal test case in a few days (I've been busy with RL
> issues).
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