Re: Digital bit perfect ouptut with ALSA

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On Tue, 16 Dec 2008, Sergei Steshenko wrote:

> On Mon, 15 Dec 2008 23:40:11 +0200
> Sergei Steshenko <steshenko_sergei@xxxxxxx> wrote:
>
>> On Mon, 15 Dec 2008 23:17:11 +0200
>> Sergei Steshenko <steshenko_sergei@xxxxxxx> wrote:
>>
>>> On Mon, 15 Dec 2008 12:56:28 -0800 (PST)
>>> Bill Unruh <unruh@xxxxxxxxxxxxxx> wrote:
>>>
>>>
>>>>
>>>> What kind of resampling does ALSA do these days-- linear interpolation ( which
>>>> is fast and does not have any delay, but introduces loads and loads of
>>>> distortion and noise) or what?
>>>
>>>
>>> You may choose. There is now sinc interpolator with low, medium and high
>>> quality. You need to instal a plugin - comes with ALSA.

The problem I believe is that this has large latency. And this is NOT the
sample rate conversion that alsa uses by default (eg if it is fed a 44.1 file
when the card only does 48K. ) I thought the default was simple linear
interpolation but am not sure anymore.


>>>
>>> One line has to be added to a configuration file - I do not remember the
>>> details.
>>>
>>> Regards,
>>>   Sergei.
>>>
>>>
>>> ------------------------------------------------------------------------------
>>
>> The plugin: alsa-plugins-samplerate.
>>
>> --Sergei.
>>
>
> And setup info for it:
>
> /usr/share/doc/packages/alsa-plugins-samplerate/samplerate.txt :
>
> "
> Rate Converter Plugin Using Libsamplerate
> =========================================
>
> The plugin in rate subdirectory is an external rate converter using
> libsamplerate by Erik de Castro Lopo.  You can use this rate converter
> plugin by defining a rate PCM with "converter" parameter, such as:
>
>        pcm.my_rate {
>                type rate
>                slave.pcm "hw"
>                converter "samplerate"
>        }
>
> The plug plugin has also a similar field, "rate_converter".
>
> Or, more easily, define a global variable "defaults.pcm.rate_converter",
> which is used as the default converter type by plug and rate plugins:
>
>        defaults.pcm.rate_converter "samplerate"
>
> Write the above in your ~/.asoundrc or /etc/asound.conf.
>
> The following converter types are available:
>
>  - samplerate_best     Use SRC_SINC_BEST_QUALITY
>  - samplerate_medium   Use SRC_SINC_MEDIUM_QUALITY
>  - samplerate          Use SRC_SINC_FASTEST
>  - samplerate_order    Use SRC_ZERO_ORDER_HOLD
>  - samplerate_linear   Use SRC_LINEAR
>
> ".
>
> In simple English, one line, e.g.:
>
>
> defaults.pcm.rate_converter "samplerate_best"
>

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