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rtsp-media: do not unblock on unsuspend

Browse source 
rtsp_media_unsuspend() is called from handle_play_request()
before sending the play response. Unblocking the streams here
was causing data to be sent out before the client was ready
to handle it, with obvious side effects such as initial packets
getting discarded, causing decoding errors.

Instead we can simply let the media streams be unblocked when
the state of the media is set to PLAYING, which occurs after
sending the play response.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
This commit is contained in:
Mathieu Duponchelle 2020-08-25 16:10:36 +02:00 committed by GStreamer Merge Bot
parent 17edff4926
commit 883ddc72bb
1 changed files with 6 additions and 13 deletions
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11
gst/rtsp-server/rtsp-media.c
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@ -4582,21 +4582,14 @@ default_unsuspend (GstRTSPMedia * media)
if (gst_rtsp_media_is_receive_only (media))
break;
if (media_streams_blocking (media)) {
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
/* at this point the media pipeline has been updated and contain all
* specific transport parts: all active streams contain at least one sink
* element and it's safe to unblock all blocked streams */
media_streams_set_blocked (media, FALSE);
} else {
/* streams are not blocked and media is suspended from PAUSED */
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
}
g_rec_mutex_unlock (&priv->state_lock);
if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
g_rec_mutex_lock (&priv->state_lock);
goto preroll_failed;
}
g_rec_mutex_lock (&priv->state_lock);
}
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
break;
case GST_RTSP_SUSPEND_MODE_PAUSE:
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);