[gst-embedded] noise and stuttering

[Dennis Fleming]

First off: Thank to you and Zhoa-Lang for getting back so quickly.  I'm so busy I forgot my manners.

Testing to find the parameters I have I used decodebin, but in the program itself uses playbin with the same effect.   The only variation is that I set the sink property to alsasink since that seems the only way to set buffer-time and latency-time properties.  Also, it seems counter-intuitive to me that an uncompressed WAV file should have problems keeping up while MP3s with the same sampling frequency and word size have none.  And yet the artifacts are indicative of dropped buffers.  

Dennis



----- Original Message ----
From: Thijs Vermeir <thijsvermeir at gmail.com>
To: Dennis Fleming <arsantiqua at sbcglobal.net>
Cc: gstreamer-embedded at lists.sourceforge.net
Sent: Tuesday, July 29, 2008 3:59:30 PM
Subject: Re: [gst-embedded] noise and stuttering

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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