Hi Keith,
thanks for your comments.
keith preston wrote:
> If you want gstreamer to be efficent in an embedded device you really have
> to crank up the buffers. This can be done by changing the "buffer-time"
> and "latency-time" on the pulsesink. I use this instead of an alsasink.
> I generally use times like 180ms for buffer and 60ms for latency. The
> properties come from the base class audio ring buffer that the pulse sink
> derives from. I believe the alsasink is the same way.
I set the settings with gconftool:
gconftool-2 -t string --set /system/gstreamer/0.10/default/audiosink
'pulsesink buffer-time=180 latency-time=60'
and this sounds better now. clicks are gone for most of the time. I am
not really convinced though of gstreamer. It uses roughly 40% of cpu
when running on the XO and playing back an ogg file. I put together a
simple pa client which uses liboog and this one consumes about 18%. I am
aware that gstreamer has got a lot of features but in such an
environment this is probably not needed. Anyway.
> I have tried another hack that made the alsa-pulse plugin refuse to set
> small buffers. This was by changing the capabilites function of the alsa
> plugin to say that there was a minimum frag size of like 16kb or something.
> This helped for native alsa applications.
>
> The biggest problem you will find with running it on the XO, (i've said
> this
> before on the list) is resampling, mixing, and software volume.
> Resampling has a little hope as there is a decent fixed point library from
> speex that is close to being done and released. It's not hard to
> integrate.
Indeed - the problem about the resampling performance was one thing I
have found out in my tests as well. I have been reading through the old
thread about the resample library and will give it a go.
> As for mixing and software volume, there need to be a lot of change. I am
> currently using a library by my processor vendor for all three of these
> areas.
On the last thread there were discussions about those changes - is this
still on the list of future changes?
> A few hints for lowering CPU usage Use autoload for the alsasink with a
> quick timeout. Pulse will use 0% when no audio is playing. Use large
> fragment sizes, my alsasink works in 6kb increments right now.
Using autoload with a quick timeout is a good one ( You mean this
parameter I guess --module-idle-time=sec). Have been playing around with
the fragment sizes as well.
Best,
Simon
>
> On 7/23/07, Simon Schamijer <simon at schampijer.de> wrote:
>>
>> Hi,
>>
>> I have put up a site where I posted my tests on using pulseaudio within
>> the OLPC environment.
>>
>> http://wiki.laptop.org/go/Pulseaudio
>>
>> Please feel free to add and comment.
>>
>> What I am a bit puzzled about is the gstreamer section. When using alsa
>> or native clients I can run pulseaudio with sr=44100Hz on the XO.
>> When it comes to use csound, gstreamer or the sdl-mixer I have to go
>> down to 22kHz and you have to use large buffers in both csound and
>> sdl-mixer to avoid clicking. This seams not to help with gstreamer. When
>> gstreamer is connected you have mostly a click at the beginning of the
>> playback.
>>
>> I changed the fragment settings of the ALSA module of pulseaudio and
>> tried to change the buffer settings in gstreamer as well.
>>
>> gst-launch-0.10 filesrc location=/home/olpc/guitcello.ogg ! oggdemux !
>> vorbisdec ! audioconvert ! 'buffer-size=2048'
>>
>> add-autoload-sink output module-alsa-sink device=plughw:0,0 rate=22050
>> sink_name=output fragments=12 fragment_size=1024
>>
>> No luck so far. More details on the tests can be found on the wiki side.
>> Maybe someone here has an idea what else I could try.
>>
>> Best,
>> Simon
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>> pulseaudio-discuss at mail.0pointer.de
>> https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
>>
>
>
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