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rtsp-media: do not unblock on unsuspend
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>
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parent
17edff4926
commit
883ddc72bb
1 changed files with 6 additions and 13 deletions
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@ -4582,21 +4582,14 @@ default_unsuspend (GstRTSPMedia * media)
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if (gst_rtsp_media_is_receive_only (media))
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break;
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if (media_streams_blocking (media)) {
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gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
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/* at this point the media pipeline has been updated and contain all
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* specific transport parts: all active streams contain at least one sink
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* element and it's safe to unblock all blocked streams */
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media_streams_set_blocked (media, FALSE);
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} else {
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/* streams are not blocked and media is suspended from PAUSED */
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gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
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}
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g_rec_mutex_unlock (&priv->state_lock);
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if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
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g_rec_mutex_unlock (&priv->state_lock);
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if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
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g_rec_mutex_lock (&priv->state_lock);
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goto preroll_failed;
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}
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g_rec_mutex_lock (&priv->state_lock);
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goto preroll_failed;
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}
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g_rec_mutex_lock (&priv->state_lock);
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gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
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break;
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case GST_RTSP_SUSPEND_MODE_PAUSE:
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gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
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