Video corruption in rtsp server with appsink->appsrc

[Serj TorresSoldado]

I have just noticed this happens with the gst-rtsp-server examples as well.

I am using the 1.6 branch.

On 28 March 2016 at 21:50, Serj TorresSoldado <torres.soldado at gmail.com>
wrote:

> Hi All,
>
>  I am doing a sink -> src push. The sink is in a separate pipeline and the
> src is created by gst-rtsp-server when creating the media pipeline.
>
>  I have tried both using a pull (need-data signal) and push (new-sample)
> methods and the result is the same.
>
>  I am copying the buffer from the sink to the source. I am setting the PTS
> on the copied buffer otherwise after the first client disconnects I am
> unable to connect again.
>
>  GST_BUFFER_PTS(bufcpy) = client->timestamp_;
>  client->timestamp_ += GST_BUFFER_DURATION(bufcpy);
>  gst_app_src_push_buffer(client->appsrc_, bufcpy);
>
>  When cofiguring the client I have "played" with the following appsrc
> properties but they don't seem to make a difference:
>
>   gst_util_set_object_arg(reinterpret_cast<GObject*>(appsrc.Get()),
> "format", "time");
>   g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "block", false,
> nullptr);
>   g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "is-live", true,
> nullptr);
>   g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "do-timestamp",
> true, nullptr);
> //  g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "stream-type",
> "random-access", nullptr);
> //  g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "max-bytes",
> 1000, nullptr);
>
> Any help would be awesome, thanks.
>
> Serj
>
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