Hi Felicia,
I still don't quite understand why you need to make 120 ms a special
case, rather than extend the code that already handles 40 ms and 60 ms.
Cheers,
Jean-Marc
On 06/01/2016 12:58 PM, Felicia Lim wrote:> Hi all,
>
> I've just realized that there's a better and simpler way of doing
this
> which ensures that analysis and selection of the mode/bandwidth etc is
> done on the whole 120 ms frame. I believe that the logic and threshold
> values for making these decisions are still valid for 120 ms, but I
> might be missing something? Please find my updated patches attached.
>
> Thanks,
> Felicia
>
>
>
> On Tue, May 31, 2016 at 7:05 PM Felicia Lim <flim at google.com
> <mailto:flim at google.com>> wrote:
>
> Hi all,
>
> We (WebRTC/Google) would like to extend Opus to natively support 120
> ms encoding instead of relying on repacketization as a post
> processing step. This is to ensure that a valid 120 ms packet is
> always available. I've attached a couple of patches to add this to
> opus_encoder(), based on the internal repacketization process
> carried out by 60 ms CELT. We intend to extend this later for the
> multistream encoder as well. The first patch refactors out the
> internal subframe encoding and repacketizing, and the second patch
> actually adds the 120 ms support.
>
> Any thoughts would be appreciated.
>
> Thanks,
> Felicia
>
>
>
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