[Speex-dev] FFT Resampler

[Alexander Chemeris]

Hi,

Here are some questions from user point of view. :)

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote:
>  I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz,
> and there aren't any obvious artifacts. I also did a 16=>16 test, and the
> results are delayed by 10ms and within +/- 1 (basically, rounding errors
> from the FFT).

Do you plan to use it in VoIP environment? If so, 10ms of additional latency
is a big cost for resampling. Do you think there are any ways to reduce it?

>  +/** Get the latency in input samples introduced by the resampler.
>  + * @param st Resampler state
>  + * @return Latency in input samples.
>  + */
>  +int
> speex_block_resampler_get_input_latency(SpeexBlockResamplerState
> *st);
>  +
>  +/** Get the latency in output samples introduced by the resampler.
>  + * @param st Resampler state
>  + * @return Latency in output samples.
>  + */
>  +int
> speex_block_resampler_get_output_latency(SpeexBlockResamplerState
> *st);

What is input and output latency? As a user, I think there is only one latency,
latency between data I passed to resampler and data I've got from it.
I suppose there may be some internal idea behind this division of latency,
but is end user interested in it?


-- 
Regards,
Alexander Chemeris.

SIPez LLC.
SIP VoIP, IM and Presence Consulting
http://www.SIPez.com
tel: +1 (617) 273-4000