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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-06-26 18:00:49 +00:00
jitterbuffer: refactor handle sync code
Move the code that combines the last SR packet and the current jitterbuffer sync values into a sync structure, into its own function. We want to reuse this bit later.
This commit is contained in:
Wim Taymans
parent
87f7d6b9bf
commit
af055d9574
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@ -179,6 +179,8 @@ struct _GstRtpJitterBufferPrivate
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/* the latency of the upstream peer, we have to take this into account when
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* synchronizing the buffers. */
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GstClockTime peer_latency;
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guint64 ext_rtptime;
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GstBuffer *last_sr;
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/* some accounting */
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guint64 num_late;
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@ -976,6 +978,7 @@ gst_rtp_jitter_buffer_change_state (GstElement * element,
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ret = GST_STATE_CHANGE_NO_PREROLL;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_buffer_replace (&priv->last_sr, NULL);
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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@ -1983,6 +1986,90 @@ pause:
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}
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}
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/* collect the info form the lastest RTCP packet and the jittebuffer sync, do
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* some sanity checks and then emit the handle-sync signal with the parameters.
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* This function must be called with the LOCK */
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static void
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do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
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{
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GstRtpJitterBufferPrivate *priv;
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guint64 base_rtptime, base_time;
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guint32 clock_rate;
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guint64 last_rtptime;
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guint64 clock_base;
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guint64 ext_rtptime, diff;
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gboolean drop = FALSE;
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priv = jitterbuffer->priv;
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if (priv->last_sr == NULL) {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping, no SR RTCP");
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return;
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}
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/* get the last values from the jitterbuffer */
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rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
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&clock_rate, &last_rtptime);
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clock_base = priv->clock_base;
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ext_rtptime = priv->ext_rtptime;
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GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
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G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
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", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
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ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
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if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
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drop = TRUE;
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} else {
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/* we can't accept anything that happened before we did the last resync */
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if (base_rtptime > ext_rtptime) {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
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drop = TRUE;
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} else {
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/* the SR RTP timestamp must be something close to what we last observed
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* in the jitterbuffer */
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if (ext_rtptime > last_rtptime) {
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/* check how far ahead it is to our RTP timestamps */
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diff = ext_rtptime - last_rtptime;
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/* if bigger than 1 second, we drop it */
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if (diff > clock_rate) {
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GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
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/* should drop this, but some RTSP servers end up with bogus
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* way too ahead RTCP packet when repeated PAUSE/PLAY,
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* so still trigger rptbin sync but invalidate RTCP data
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* (sync might use other methods) */
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ext_rtptime = -1;
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}
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GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
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G_GUINT64_FORMAT, last_rtptime, diff);
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}
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}
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}
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if (!drop) {
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GstStructure *s;
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s = gst_structure_new ("application/x-rtp-sync",
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"base-rtptime", G_TYPE_UINT64, base_rtptime,
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"base-time", G_TYPE_UINT64, base_time,
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"clock-rate", G_TYPE_UINT, clock_rate,
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"clock-base", G_TYPE_UINT64, clock_base,
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"sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
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"sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
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GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
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JBUF_UNLOCK (priv);
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g_signal_emit (jitterbuffer,
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gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
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JBUF_LOCK (priv);
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gst_structure_free (s);
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} else {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
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}
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}
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static GstFlowReturn
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gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
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GstBuffer * buffer)
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@ -1990,16 +2077,11 @@ gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
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GstRtpJitterBuffer *jitterbuffer;
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GstRtpJitterBufferPrivate *priv;
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GstFlowReturn ret = GST_FLOW_OK;
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guint64 base_rtptime, base_time;
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guint32 clock_rate;
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guint64 last_rtptime;
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guint32 ssrc;
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GstRTCPPacket packet;
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guint64 ext_rtptime, diff;
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guint64 ext_rtptime;
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guint32 rtptime;
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gboolean drop = FALSE;
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GstRTCPBuffer rtcp = { NULL, };
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guint64 clock_base;
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jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
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@ -2032,67 +2114,12 @@ gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
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ext_rtptime = priv->jbuf->ext_rtptime;
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ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
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/* get the last values from the jitterbuffer */
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rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
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&clock_rate, &last_rtptime);
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priv->ext_rtptime = ext_rtptime;
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gst_buffer_replace (&priv->last_sr, buffer);
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clock_base = priv->clock_base;
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GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
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G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
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", clock-base %" G_GUINT64_FORMAT,
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ext_rtptime, base_rtptime, clock_rate, clock_base);
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if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
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drop = TRUE;
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} else {
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/* we can't accept anything that happened before we did the last resync */
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if (base_rtptime > ext_rtptime) {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
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drop = TRUE;
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} else {
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/* the SR RTP timestamp must be something close to what we last observed
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* in the jitterbuffer */
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if (ext_rtptime > last_rtptime) {
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/* check how far ahead it is to our RTP timestamps */
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diff = ext_rtptime - last_rtptime;
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/* if bigger than 1 second, we drop it */
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if (diff > clock_rate) {
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GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
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/* should drop this, but some RTSP servers end up with bogus
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* way too ahead RTCP packet when repeated PAUSE/PLAY,
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* so still trigger rptbin sync but invalidate RTCP data
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* (sync might use other methods) */
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ext_rtptime = -1;
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}
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GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
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G_GUINT64_FORMAT, last_rtptime, diff);
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}
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}
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}
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do_handle_sync (jitterbuffer);
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JBUF_UNLOCK (priv);
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if (!drop) {
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GstStructure *s;
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s = gst_structure_new ("application/x-rtp-sync",
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"base-rtptime", G_TYPE_UINT64, base_rtptime,
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"base-time", G_TYPE_UINT64, base_time,
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"clock-rate", G_TYPE_UINT, clock_rate,
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"clock-base", G_TYPE_UINT64, clock_base,
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"sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
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"sr-buffer", GST_TYPE_BUFFER, buffer, NULL);
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GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
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g_signal_emit (jitterbuffer,
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gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
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gst_structure_free (s);
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} else {
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GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
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ret = GST_FLOW_OK;
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}
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done:
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gst_buffer_unref (buffer);
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