[Bug 752531] [rtspsrc/rtpjitterbuffer]: playback and seek is unstable if set buffering mode

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=752531

--- Comment #3 from Nicolas Dufresne (stormer) <nicolas.dufresne at collabora.co.uk> ---
(In reply to jihae from comment #2)
> hm.. ok. "I need to get buffering msg from the pipeline." means buffering
> msg with buffering percent should be informed to the application.

When a pipeline is live, the buffering is not in size or percent, but in time.
In fact, it match the latency time. So from the time you set the pipeline to
playing, to that time + latency, the "buffering" is uniformly distributed. You
can easily right a UI around that.

> 
> The problem is "buffer-mode" setting of rtspsrc/rtpjitterbuffer.
> if want to get buffering msg with percent, have to set "buffer-mode"
> property as 'BUFFER_MODE_BUFFER' .
> 
> 
>   static const GEnumValue buffer_modes[] = {
>     {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
>     {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
>     {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
>     {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
>     {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks",
> "synced"},
>     {0, NULL, NULL},
>   };

I'll let other comment about this mode, I still can't think of how to make this
work in a live context. Clearly it does not seem implemented well.

-- 
You are receiving this mail because:
You are the QA Contact for the bug.
You are the assignee for the bug.