----- Original Message ----
From: Thijs Vermeir <email@example.com>
To: Dennis Fleming <firstname.lastname@example.org>
Sent: Tuesday, July 29, 2008 3:59:30 PM
Subject: Re: [gst-embedded] noise and stuttering
On Tue, Jul 29, 2008 at 5:43 PM, Dennis Fleming
> 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. 
> 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?
> ----- Original Message ----
> From: Thijs Vermeir <email@example.com
> To: Zhao Liang-E3423C <E3423C@motorola.com
> Cc: Dennis Fleming <firstname.lastname@example.org
> Sent: Tuesday, July 29, 2008 2:46:42 AM
> Subject: Re: [gst-embedded] noise and stuttering
> On Tue, Jul 29, 2008 at 11:15 AM, Zhao
>> 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.
>> 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
>> Zhao liang
>> From: email@example.com
] On Behalf Of
>> Dennis Fleming
>> Sent: Tuesday, July 29, 2008 4:37 AM
>> To: firstname.lastname@example.org
>> Subject: [gst-embedded] noise and stuttering
>> I'm trying to create an audio player on an IMX31 target and I've
>> discrepancy in the output of various formats. If I send MP3 data I have
>> 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
>> account for this difference?
>> Linux 22.214.171.124
>> gstreamer 0.10.17 (open-embedded)
>> gst-launch filesrc location=<file> ! decodebin ! alsasink
>> This SF.Net email is sponsored by the Moblin Your Move Developer's
>> Build the coolest Linux based applications with Moblin SDK & win great
>> Grand prize is a trip for two to an Open Source event anywhere in the
>> Gstreamer-embedded mailing list