[Speex-dev] Audio Speed Variability

[Jean-Marc Valin]

Actually, Speex now includes a resampler. I suggest you use that.
There's even a parameter that allows to control the CPU/quality tradeoff.

	Jean-Marc

John Miles wrote:
> I don't know about the input side; I have personally only experienced being
> bitten by the output resampler.  But it seems like a safe assumption that
> yes, the input side is equally broken.
> 
> Any resampling code found on the 'net should be suitable as long as it
> sounds good, doesn't take too much CPU, and is compatible with your
> product's licensing/distribution terms.  There are quite a few options out
> there, but I haven't investigated any of them.  It's probably reasonable to
> replicate the sample points yourself and run the result through an FIR or
> IIR low-pass filter kernel that's hardwired to bandlimit the resulting data
> for the rate your codec uses.  E.g., 16000 Hz to 44100 Hz would involve
> making two or three copies of each sample point as necessary, then running
> an FIR or IIR filter calculated for 0.36 Nyquist.
> 
> If you have only one input and one output stream to deal with, both with
> fixed rates, it's hard to find a resampler that *won't* work.
> 
> -- john
> 
>> -----Original Message-----
>> From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org]On
>> Behalf Of James Stanton
>> Sent: Thursday, October 04, 2007 3:14 PM
>> To: speex-dev at xiph.org
>> Subject: Re: [Speex-dev] Audio Speed Variability
>>
>>
>> John,
>>
>> Thanks for the reply!  You mentioned output sample rates should be 44100
>> or 48000, should I worry about input (Mic) Sample rates as well?
>> (Currently I was requesting the sample rate on both ends to be 16000
>> samplesPerSecond, for ease of passing into the codec)   Also, do you
>> recommend any particular resampler that I should use, or are any of the
>> ones out there probably okay, or should I just write my own?  Thanks
>> again for your help!
>>
>> Jamie
>>
>> John Miles wrote:
>>
>>>> -----Original Message-----
>>>> From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at xiph.org]On
>>>> Behalf Of James Stanton
>>>> Sent: Thursday, October 04, 2007 12:53 PM
>>>> To: speex-dev at xiph.org
>>>> Subject: [Speex-dev] Audio Speed Variability
>>>>
>>>>
>>>> I have a video conference like application that I've been working on for
>>>> a while now, and a recent change is causing some odd problems, and I was
>>>> wondering if anyone else had seen problems like this....
>>>>
>>>>
>>>
>>> Short answer: don't use output sample rates other than 44100 or 48000.
>>>
>>> Longer answer: Sound chips usually run at one of those rates,
>> often either.
>>> Those rates are more or less guaranteed to work properly.  Most
>> chips don't
>>> support other rates directly; a software resampler in the driver is used
>>> instead.  Unfortunately, Microsoft released a horribly-broken reference
>>> resampler implementation to sound hardware OEMs a few years ago,
>> and many of
>>> them still use it.  On their sound cards, if you ask for 11025 Hz, for
>>> example you're likely to get 11100 Hz or something similarly-imprecise.
>>> That obviously causes cumulative latency/slippage problems.
>>>
>>> Bottom line: voice codec applications that need to work at lower rates
>>> really need to resample to 44.1K or 48K themselves in order to
>> work robustly
>>> across all hardware platforms.  Neither MS nor sound-hardware OEMs have
>>> shown the slightest interest in fixing this bug, so that's just
>> the way it
>>> goes.
>>>
>>> -- john
>>>
>>> _______________________________________________
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>>> Speex-dev at xiph.org
>>> http://lists.xiph.org/mailman/listinfo/speex-dev
>>>
>>>
>>
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