[FFmpeg-devel] [PATCH] fix speex sample

[Baptiste Coudurier]

Hi Michael,

On 4/2/2009 8:46 PM, Michael Niedermayer wrote:
> On Thu, Apr 02, 2009 at 08:08:41PM -0700, Baptiste Coudurier wrote:
>> On 4/2/2009 7:11 PM, Michael Niedermayer wrote:
>>> On Thu, Apr 02, 2009 at 06:09:15PM -0700, Baptiste Coudurier wrote:
>>>> Michael Niedermayer wrote:
>>> [...]
>>>
>>>>>> Then I guess you noticed how many bugs are rotting in roundup ? This is
>>>>>> becoming a nightmare and is really frightening. Nobody jumps in, and
>>>>>> this was proved by the last 2 attempts to organize a bug fixing weekend.
>>>>> i know and agree, you also remember i did occassionally in the past fix a
>>>>> bunch of bugs but ATM i lack the motivation, i know thats my problem and
>>>>> i should go and bang my head against the wall ;)
>>>> Well, yes, but it's not only your problem, I believe it is our problem,
>>>> and it would be nice if everybody could participate in the effort.
>>> fine but i have to check this with the insurance first, if they cover damage
>>> from a bunch of madman banging their heads against the walls
>>>
>>>
>>>> I was motivated yesterday night.
>>>>
>>>>>> Now I do care and I proved it, so I claim that yes some changes my
>>>>>> introduce regression but these will be fixed quickly, I engage myself.
>>>>> i think the proposed fix is wrong and a correct one will not introduce a
>>>>> regression
>>>> Well, I still don't really see how and why, I've explained more below.
>>> iam not talking about "SoundRate UB[2]" but that:
>>>             else if(!strcmp(key, "audiosamplerate") && acodec && num_val >= 0) {
>>>                 //some tools, like FLVTool2, write consistently approximate metadata sample rates
>>>                 if (!acodec->sample_rate) {
>>>                     switch((int)num_val) {
>>>                         case 44000: acodec->sample_rate = 44100  ; break;
>>>                         case 22000: acodec->sample_rate = 22050  ; break;
>>>                         case 11000: acodec->sample_rate = 11025  ; break;
>>>                         case 5000 : acodec->sample_rate = 5512   ; break;
>>>                         default   : acodec->sample_rate = num_val;
>> Yes, however I'm sorry, I don't see how this relates to the patch.
>>
>> The patch makes the demuxer always set speex and nellymoser to their
>> correct _codec_ value IMHO, this "audiosamplerate" is a metadata thing
>> according to the specs, defined with onMetadata mechanism.
> 
> I agree but then there would be no point to even decode audiosamplerate
> and i suspect there was some sense in this, i just dont remember it now
> 

Any update on this ? It would be nice to fix bugs ;)

Thanks Michael.

-- 
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