[gst-embedded] noise and stuttering

[Thijs Vermeir]

Hi,

On Tue, Jul 29, 2008 at 5:43 PM, Dennis Fleming
<arsantiqua at sbcglobal.net> wrote:
> The interesting thing is that uncompressed WAV files are causing the problem
> while MP3s were fixed by setting the buffer-time and latency-time to values
> smaller than found on a desktop.  What would adding a queue do to latency
> through the system?

There is no latency in this case because there are no live-sources. [1]

> Also, I suppose, that I will need to break up the
> playbin and create a pipeline myself, yes?

playbin has the queue elements on the correct location, no changes needed.
You where already using a custom pipeline, no?

Gr,

[1] http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup

>
> Dennis
>
> ----- Original Message ----
> From: Thijs Vermeir <thijsvermeir at gmail.com>
> To: Zhao Liang-E3423C <E3423C at motorola.com>
> Cc: Dennis Fleming <arsantiqua at sbcglobal.net>;
> gstreamer-embedded at lists.sourceforge.net
> Sent: Tuesday, July 29, 2008 2:46:42 AM
> Subject: Re: [gst-embedded] noise and stuttering
>
> Hi,
>
> On Tue, Jul 29, 2008 at 11:15 AM, Zhao Liang-E3423C <E3423C at motorola.com>
> wrote:
>> What's the rootcause of noise and stuttering ?
>
> Now you are using only 1 thread for all the elements and if the
> filesrc or the decoder is too slow sometimes
> you don't have time to catch up. By adding the queue you put the sink
> in another thread and now the filesrc+decoder can
> do some decoding in advance.
>
> Gr,
> Thijs
>
>>
>> For normal playback, it should not have issues. If decoder didn't drop
>> data, I think alsasink did it.
>> By gstaudiosink mechanism, it will drop data replaced with blank data when
>> data is late. I guess the rootcause is that.
>>
>> If that, I have no ideas except adding a queue before alsasink, and when
>> queue is empty, pause the pipeline, it will not cause dropout, but still
>> discontinous.
>>
>> Zhao liang
>> ________________________________
>> From: gstreamer-embedded-bounces at lists.sourceforge.net
>> [mailto:gstreamer-embedded-bounces at lists.sourceforge.net] On Behalf Of
>> Dennis Fleming
>> Sent: Tuesday, July 29, 2008 4:37 AM
>> To: gstreamer-embedded at lists.sourceforge.net
>> Subject: [gst-embedded] noise and stuttering
>>
>> I'm trying to create an audio player on an IMX31 target and I've found a
>> discrepancy in the output of various formats.  If I send MP3 data I have
>> to
>> set the buffer-time and latency-time to 10000 and 100 respectively to play
>> without severe dropouts.  However WAV files still have drop-out at a
>> consistent rate (about 1 per 10 sec).  Are there some general features I'm
>> missing or is there some guidance on the buffer-time/latency time that
>> would
>> account for this difference?
>>
>> Linux 2.6.22.19
>> gstreamer 0.10.17 (open-embedded)
>> gst-launch filesrc location=<file> ! decodebin ! alsasink
>> buffer-time=10000
>> latency-time=100
>>
>> Dennis
>>
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>