static void
rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
{
- jbuf->packets = g_queue_new ();
+ g_mutex_init (&jbuf->clock_lock);
+
+ g_queue_init (&jbuf->packets);
jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE;
rtp_jitter_buffer_reset_skew (jbuf);
if (jbuf->media_clock_synced_id)
g_signal_handler_disconnect (jbuf->media_clock,
jbuf->media_clock_synced_id);
- if (jbuf->media_clock)
+ if (jbuf->media_clock) {
+ /* Make sure to clear any clock master before releasing the clock */
+ gst_clock_set_master (jbuf->media_clock, NULL);
gst_object_unref (jbuf->media_clock);
+ }
if (jbuf->pipeline_clock)
gst_object_unref (jbuf->pipeline_clock);
- g_queue_free (jbuf->packets);
+ /* We cannot use g_queue_clear() as it would pass the wrong size to
+ * g_slice_free() which may lead to data corruption in the slice allocator.
+ */
+ rtp_jitter_buffer_flush (jbuf, NULL, NULL);
+
+ g_mutex_clear (&jbuf->clock_lock);
G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
}
return jbuf->clock_rate;
}
-/* FIXME: This is called from another thread than anything else
- * and will need some locking */
static void
media_clock_synced_cb (GstClock * clock, gboolean synced,
RTPJitterBuffer * jbuf)
{
GstClockTime internal, external;
- if (!jbuf->pipeline_clock)
- return;
-
- internal = gst_clock_get_internal_time (jbuf->media_clock);
- external = gst_clock_get_time (jbuf->pipeline_clock);
+ g_mutex_lock (&jbuf->clock_lock);
+ if (jbuf->pipeline_clock) {
+ internal = gst_clock_get_internal_time (jbuf->media_clock);
+ external = gst_clock_get_time (jbuf->pipeline_clock);
- gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
+ gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
+ }
+ g_mutex_unlock (&jbuf->clock_lock);
}
/**
rtp_jitter_buffer_set_media_clock (RTPJitterBuffer * jbuf, GstClock * clock,
guint64 clock_offset)
{
+ g_mutex_lock (&jbuf->clock_lock);
if (jbuf->media_clock) {
if (jbuf->media_clock_synced_id)
g_signal_handler_disconnect (jbuf->media_clock,
gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
}
+ g_mutex_unlock (&jbuf->clock_lock);
}
/**
* rtp_jitter_buffer_set_pipeline_clock:
* @jbuf: an #RTPJitterBuffer
- * @clock: (transfer full): pipeline #GstClock
+ * @clock: pipeline #GstClock
*
* Sets the pipeline clock
*
void
rtp_jitter_buffer_set_pipeline_clock (RTPJitterBuffer * jbuf, GstClock * clock)
{
+ g_mutex_lock (&jbuf->clock_lock);
if (jbuf->pipeline_clock)
gst_object_unref (jbuf->pipeline_clock);
- jbuf->pipeline_clock = clock;
+ jbuf->pipeline_clock = clock ? gst_object_ref (clock) : NULL;
if (jbuf->pipeline_clock && jbuf->media_clock &&
jbuf->pipeline_clock != jbuf->media_clock) {
gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
}
+ g_mutex_unlock (&jbuf->clock_lock);
+}
+
+gboolean
+rtp_jitter_buffer_get_rfc7273_sync (RTPJitterBuffer * jbuf)
+{
+ return jbuf->rfc7273_sync;
+}
+
+void
+rtp_jitter_buffer_set_rfc7273_sync (RTPJitterBuffer * jbuf,
+ gboolean rfc7273_sync)
+{
+ jbuf->rfc7273_sync = rfc7273_sync;
}
/**
jbuf->base_time = -1;
jbuf->base_rtptime = -1;
jbuf->base_extrtp = -1;
+ jbuf->media_clock_base_time = -1;
jbuf->ext_rtptime = -1;
jbuf->last_rtptime = -1;
jbuf->window_pos = 0;
jbuf->prev_send_diff = -1;
jbuf->prev_out_time = -1;
jbuf->need_resync = TRUE;
+
GST_DEBUG ("reset skew correction");
}
GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
{
jbuf->base_time = time;
+ jbuf->media_clock_base_time = -1;
jbuf->base_rtptime = gstrtptime;
jbuf->base_extrtp = ext_rtptime;
jbuf->prev_out_time = -1;
guint64 level;
/* first buffer with timestamp */
- high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
+ high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (&jbuf->packets);
while (high_buf) {
if (high_buf->dts != -1 || high_buf->pts != -1)
break;
high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
}
- low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
+ low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (&jbuf->packets);
while (low_buf) {
if (low_buf->dts != -1 || low_buf->pts != -1)
break;
* Cri : The time of the clock at the receiver for packet i
* D + ni : The jitter when receiving packet i
*
- * We see that the network delay is irrelevant here as we can elliminate D:
+ * We see that the network delay is irrelevant here as we can eliminate D:
*
* recv_diff(i) = (Cri + ni) - (Cr0 + n0))
*
* Returns: @time adjusted with the clock skew.
*/
static GstClockTime
-calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
+calculate_skew (RTPJitterBuffer * jbuf, guint64 ext_rtptime,
+ GstClockTime gstrtptime, GstClockTime time, gint gap, gboolean is_rtx)
{
- guint64 ext_rtptime;
guint64 send_diff, recv_diff;
gint64 delta;
gint64 old;
gint pos, i;
- GstClockTime gstrtptime, out_time;
+ GstClockTime out_time;
guint64 slope;
- ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
-
- if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
- return jbuf->prev_out_time;
-
- gstrtptime =
- gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
-
- /* keep track of the last extended rtptime */
- jbuf->last_rtptime = ext_rtptime;
-
- send_diff = 0;
- if (G_LIKELY (jbuf->base_rtptime != -1)) {
- /* check elapsed time in RTP units */
- if (G_LIKELY (gstrtptime >= jbuf->base_rtptime)) {
- send_diff = gstrtptime - jbuf->base_rtptime;
- } else {
- /* elapsed time at sender, timestamps can go backwards and thus be
- * smaller than our base time, schedule to take a new base time in
- * that case. */
- GST_WARNING ("backward timestamps at server, schedule resync");
- jbuf->need_resync = TRUE;
- send_diff = 0;
- }
- }
-
- /* need resync, lock on to time and gstrtptime if we can, otherwise we
- * do with the previous values */
- if (G_UNLIKELY (jbuf->need_resync && time != -1)) {
- GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
- GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (gstrtptime));
- rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
- send_diff = 0;
- }
-
- GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
- GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
- GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
- GST_TIME_ARGS (send_diff));
+ /* elapsed time at sender */
+ send_diff = gstrtptime - jbuf->base_rtptime;
/* we don't have an arrival timestamp so we can't do skew detection. we
* should still apply a timestamp based on RTP timestamp and base_time */
- if (time == -1 || jbuf->base_time == -1)
+ if (time == -1 || jbuf->base_time == -1 || is_rtx)
goto no_skew;
/* elapsed time at receiver, includes the jitter */
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
send_diff = 0;
delta = 0;
+ gap = 0;
}
+ /* only do skew calculations if we didn't have a gap. if too much time
+ * has elapsed despite there being a gap, we resynced already. */
+ if (G_UNLIKELY (gap != 0))
+ goto no_skew;
+
pos = jbuf->window_pos;
if (G_UNLIKELY (jbuf->window_filling)) {
} else {
out_time += jbuf->skew;
}
- /* check if timestamps are not going backwards, we can only check this if we
- * have a previous out time and a previous send_diff */
- if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
- /* now check for backwards timestamps */
- if (G_UNLIKELY (
- /* if the server timestamps went up and the out_time backwards */
- (send_diff > jbuf->prev_send_diff
- && out_time < jbuf->prev_out_time) ||
- /* if the server timestamps went backwards and the out_time forwards */
- (send_diff < jbuf->prev_send_diff
- && out_time > jbuf->prev_out_time) ||
- /* if the server timestamps did not change */
- send_diff == jbuf->prev_send_diff)) {
- GST_DEBUG ("backwards timestamps, using previous time");
- out_time = jbuf->prev_out_time;
- }
- }
- if (time != -1 && out_time + jbuf->delay < time) {
- /* if we are going to produce a timestamp that is later than the input
- * timestamp, we need to reset the jitterbuffer. Likely the server paused
- * temporarily */
- GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
- GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
- jbuf->delay, GST_TIME_ARGS (time));
- rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
- out_time = time;
- send_diff = 0;
- }
} else
out_time = -1;
- jbuf->prev_out_time = out_time;
- jbuf->prev_send_diff = send_diff;
-
GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
jbuf->skew, GST_TIME_ARGS (out_time));
static void
queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
{
- GQueue *queue = jbuf->packets;
+ GQueue *queue = &jbuf->packets;
/* It's more likely that the packet was inserted at the tail of the queue */
if (G_LIKELY (list)) {
queue->length++;
}
+GstClockTime
+rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
+ gboolean estimated_dts, guint32 rtptime, GstClockTime base_time,
+ gint gap, gboolean is_rtx)
+{
+ guint64 ext_rtptime;
+ GstClockTime gstrtptime, pts;
+ GstClock *media_clock, *pipeline_clock;
+ guint64 media_clock_offset;
+ gboolean rfc7273_mode;
+
+ /* rtp time jumps are checked for during skew calculation, but bypassed
+ * in other mode, so mind those here and reset jb if needed.
+ * Only reset if valid input time, which is likely for UDP input
+ * where we expect this might happen due to async thread effects
+ * (in seek and state change cycles), but not so much for TCP input */
+ if (GST_CLOCK_TIME_IS_VALID (dts) && !estimated_dts &&
+ jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
+ jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
+ GstClockTime ext_rtptime = jbuf->ext_rtptime;
+
+ ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
+ if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
+ ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
+ if (!is_rtx) {
+ /* reset even if we don't have valid incoming time;
+ * still better than producing possibly very bogus output timestamp */
+ GST_WARNING ("rtp delta too big, reset skew");
+ rtp_jitter_buffer_reset_skew (jbuf);
+ } else {
+ GST_WARNING ("rtp delta too big: ignore rtx packet");
+ media_clock = NULL;
+ pipeline_clock = NULL;
+ pts = GST_CLOCK_TIME_NONE;
+ goto done;
+ }
+ }
+ }
+
+ /* Return the last time if we got the same RTP timestamp again */
+ ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
+ if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime) {
+ return jbuf->prev_out_time;
+ }
+
+ /* keep track of the last extended rtptime */
+ jbuf->last_rtptime = ext_rtptime;
+
+ g_mutex_lock (&jbuf->clock_lock);
+ media_clock = jbuf->media_clock ? gst_object_ref (jbuf->media_clock) : NULL;
+ pipeline_clock =
+ jbuf->pipeline_clock ? gst_object_ref (jbuf->pipeline_clock) : NULL;
+ media_clock_offset = jbuf->media_clock_offset;
+ g_mutex_unlock (&jbuf->clock_lock);
+
+ gstrtptime =
+ gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
+
+ if (G_LIKELY (jbuf->base_rtptime != -1)) {
+ /* check elapsed time in RTP units */
+ if (gstrtptime < jbuf->base_rtptime) {
+ if (!is_rtx) {
+ /* elapsed time at sender, timestamps can go backwards and thus be
+ * smaller than our base time, schedule to take a new base time in
+ * that case. */
+ GST_WARNING ("backward timestamps at server, schedule resync");
+ jbuf->need_resync = TRUE;
+ } else {
+ GST_WARNING ("backward timestamps: ignore rtx packet");
+ pts = GST_CLOCK_TIME_NONE;
+ goto done;
+ }
+ }
+ }
+
+ switch (jbuf->mode) {
+ case RTP_JITTER_BUFFER_MODE_NONE:
+ case RTP_JITTER_BUFFER_MODE_BUFFER:
+ /* send 0 as the first timestamp and -1 for the other ones. This will
+ * interpolate them from the RTP timestamps with a 0 origin. In buffering
+ * mode we will adjust the outgoing timestamps according to the amount of
+ * time we spent buffering. */
+ if (jbuf->base_time == -1)
+ dts = 0;
+ else
+ dts = -1;
+ break;
+ case RTP_JITTER_BUFFER_MODE_SYNCED:
+ /* synchronized clocks, take first timestamp as base, use RTP timestamps
+ * to interpolate */
+ if (jbuf->base_time != -1 && !jbuf->need_resync)
+ dts = -1;
+ break;
+ case RTP_JITTER_BUFFER_MODE_SLAVE:
+ default:
+ break;
+ }
+
+ /* need resync, lock on to time and gstrtptime if we can, otherwise we
+ * do with the previous values */
+ if (G_UNLIKELY (jbuf->need_resync && dts != -1)) {
+ if (is_rtx) {
+ GST_DEBUG ("not resyncing on rtx packet, discard");
+ pts = GST_CLOCK_TIME_NONE;
+ goto done;
+ }
+ GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (dts), GST_TIME_ARGS (gstrtptime));
+ rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, FALSE);
+ }
+
+ GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
+ GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
+ GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
+ GST_TIME_ARGS (gstrtptime - jbuf->base_rtptime));
+
+ rfc7273_mode = media_clock && pipeline_clock
+ && gst_clock_is_synced (media_clock);
+
+ if (rfc7273_mode && jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
+ && (media_clock_offset == -1 || !jbuf->rfc7273_sync)) {
+ GstClockTime internal, external;
+ GstClockTime rate_num, rate_denom;
+ GstClockTime nsrtptimediff, rtpntptime, rtpsystime;
+
+ gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
+ &rate_denom);
+
+ /* Slave to the RFC7273 media clock instead of trying to estimate it
+ * based on receive times and RTP timestamps */
+
+ if (jbuf->media_clock_base_time == -1) {
+ if (jbuf->base_time != -1) {
+ jbuf->media_clock_base_time =
+ gst_clock_unadjust_with_calibration (media_clock,
+ jbuf->base_time + base_time, internal, external, rate_num,
+ rate_denom);
+ } else {
+ if (dts != -1)
+ jbuf->media_clock_base_time =
+ gst_clock_unadjust_with_calibration (media_clock, dts + base_time,
+ internal, external, rate_num, rate_denom);
+ else
+ jbuf->media_clock_base_time =
+ gst_clock_get_internal_time (media_clock);
+ jbuf->base_rtptime = gstrtptime;
+ }
+ }
+
+ if (gstrtptime > jbuf->base_rtptime)
+ nsrtptimediff = gstrtptime - jbuf->base_rtptime;
+ else
+ nsrtptimediff = 0;
+
+ rtpntptime = nsrtptimediff + jbuf->media_clock_base_time;
+
+ rtpsystime =
+ gst_clock_adjust_with_calibration (media_clock, rtpntptime, internal,
+ external, rate_num, rate_denom);
+
+ if (rtpsystime > base_time)
+ pts = rtpsystime - base_time;
+ else
+ pts = 0;
+
+ GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
+ } else if (rfc7273_mode && (jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
+ || jbuf->mode == RTP_JITTER_BUFFER_MODE_SYNCED)
+ && media_clock_offset != -1 && jbuf->rfc7273_sync) {
+ GstClockTime ntptime, rtptime_tmp;
+ GstClockTime ntprtptime, rtpsystime;
+ GstClockTime internal, external;
+ GstClockTime rate_num, rate_denom;
+
+ /* Don't do any of the dts related adjustments further down */
+ dts = -1;
+
+ /* Calculate the actual clock time on the sender side based on the
+ * RFC7273 clock and convert it to our pipeline clock
+ */
+
+ gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
+ &rate_denom);
+
+ ntptime = gst_clock_get_internal_time (media_clock);
+
+ ntprtptime = gst_util_uint64_scale (ntptime, jbuf->clock_rate, GST_SECOND);
+ ntprtptime += media_clock_offset;
+ ntprtptime &= 0xffffffff;
+
+ rtptime_tmp = rtptime;
+ /* Check for wraparounds, we assume that the diff between current RTP
+ * timestamp and current media clock time can't be bigger than
+ * 2**31 clock units */
+ if (ntprtptime > rtptime_tmp && ntprtptime - rtptime_tmp >= 0x80000000)
+ rtptime_tmp += G_GUINT64_CONSTANT (0x100000000);
+ else if (rtptime_tmp > ntprtptime && rtptime_tmp - ntprtptime >= 0x80000000)
+ ntprtptime += G_GUINT64_CONSTANT (0x100000000);
+
+ if (ntprtptime > rtptime_tmp)
+ ntptime -=
+ gst_util_uint64_scale (ntprtptime - rtptime_tmp, GST_SECOND,
+ jbuf->clock_rate);
+ else
+ ntptime +=
+ gst_util_uint64_scale (rtptime_tmp - ntprtptime, GST_SECOND,
+ jbuf->clock_rate);
+
+ rtpsystime =
+ gst_clock_adjust_with_calibration (media_clock, ntptime, internal,
+ external, rate_num, rate_denom);
+ /* All this assumes that the pipeline has enough additional
+ * latency to cover for the network delay */
+ if (rtpsystime > base_time)
+ pts = rtpsystime - base_time;
+ else
+ pts = 0;
+
+ GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
+ } else {
+ /* If we used the RFC7273 clock before and not anymore,
+ * we need to resync it later again */
+ jbuf->media_clock_base_time = -1;
+
+ /* do skew calculation by measuring the difference between rtptime and the
+ * receive dts, this function will return the skew corrected rtptime. */
+ pts = calculate_skew (jbuf, ext_rtptime, gstrtptime, dts, gap, is_rtx);
+ }
+
+ /* check if timestamps are not going backwards, we can only check this if we
+ * have a previous out time and a previous send_diff */
+ if (G_LIKELY (pts != -1 && jbuf->prev_out_time != -1
+ && jbuf->prev_send_diff != -1)) {
+ /* now check for backwards timestamps */
+ if (G_UNLIKELY (
+ /* if the server timestamps went up and the out_time backwards */
+ (gstrtptime - jbuf->base_rtptime > jbuf->prev_send_diff
+ && pts < jbuf->prev_out_time) ||
+ /* if the server timestamps went backwards and the out_time forwards */
+ (gstrtptime - jbuf->base_rtptime < jbuf->prev_send_diff
+ && pts > jbuf->prev_out_time) ||
+ /* if the server timestamps did not change */
+ gstrtptime - jbuf->base_rtptime == jbuf->prev_send_diff)) {
+ GST_DEBUG ("backwards timestamps, using previous time");
+ pts = jbuf->prev_out_time;
+ }
+ }
+
+ if (gap == 0 && dts != -1 && pts + jbuf->delay < dts) {
+ /* if we are going to produce a timestamp that is later than the input
+ * timestamp, we need to reset the jitterbuffer. Likely the server paused
+ * temporarily */
+ GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
+ GST_TIME_FORMAT ", reset jitterbuffer and discard", GST_TIME_ARGS (pts),
+ jbuf->delay, GST_TIME_ARGS (dts));
+ rtp_jitter_buffer_reset_skew (jbuf);
+ rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, TRUE);
+ pts = dts;
+ }
+
+ jbuf->prev_out_time = pts;
+ jbuf->prev_send_diff = gstrtptime - jbuf->base_rtptime;
+
+done:
+ if (media_clock)
+ gst_object_unref (media_clock);
+ if (pipeline_clock)
+ gst_object_unref (pipeline_clock);
+
+ return pts;
+}
+
+
/**
* rtp_jitter_buffer_insert:
* @jbuf: an #RTPJitterBuffer
*
* Returns: %FALSE if a packet with the same number already existed.
*/
-gboolean
+static gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
gboolean * head, gint * percent)
{
GList *list, *event = NULL;
- guint32 rtptime;
guint16 seqnum;
- GstClockTime dts;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (item != NULL, FALSE);
- list = jbuf->packets->tail;
+ list = jbuf->packets.tail;
/* no seqnum, simply append then */
if (item->seqnum == -1)
if (event)
list = event;
- dts = item->dts;
- if (item->rtptime == -1)
- goto append;
-
- rtptime = item->rtptime;
-
- /* rtp time jumps are checked for during skew calculation, but bypassed
- * in other mode, so mind those here and reset jb if needed.
- * Only reset if valid input time, which is likely for UDP input
- * where we expect this might happen due to async thread effects
- * (in seek and state change cycles), but not so much for TCP input */
- if (GST_CLOCK_TIME_IS_VALID (dts) &&
- jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
- jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
- GstClockTime ext_rtptime = jbuf->ext_rtptime;
-
- ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
- if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
- ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
- /* reset even if we don't have valid incoming time;
- * still better than producing possibly very bogus output timestamp */
- GST_WARNING ("rtp delta too big, reset skew");
- rtp_jitter_buffer_reset_skew (jbuf);
- }
- }
-
- switch (jbuf->mode) {
- case RTP_JITTER_BUFFER_MODE_NONE:
- case RTP_JITTER_BUFFER_MODE_BUFFER:
- /* send 0 as the first timestamp and -1 for the other ones. This will
- * interpolate them from the RTP timestamps with a 0 origin. In buffering
- * mode we will adjust the outgoing timestamps according to the amount of
- * time we spent buffering. */
- if (jbuf->base_time == -1)
- dts = 0;
- else
- dts = -1;
- break;
- case RTP_JITTER_BUFFER_MODE_SYNCED:
- /* synchronized clocks, take first timestamp as base, use RTP timestamps
- * to interpolate */
- if (jbuf->base_time != -1)
- dts = -1;
- break;
- case RTP_JITTER_BUFFER_MODE_SLAVE:
- default:
- break;
- }
-
- if (gst_clock_is_synced (jbuf->media_clock)) {
- GstClockTime ntptime, systime;
- GstClockTime ntprtptime, rtpsystime;
-
- ntptime = gst_clock_get_internal_time (jbuf->media_clock);
- systime = gst_clock_get_time (jbuf->pipeline_clock);
- g_print ("ntp %" GST_TIME_FORMAT " sys %" GST_TIME_FORMAT "\n",
- GST_TIME_ARGS (ntptime), GST_TIME_ARGS (systime));
-
- ntprtptime = gst_util_uint64_scale (ntptime, jbuf->clock_rate, GST_SECOND);
- ntprtptime += jbuf->media_clock_offset;
- ntprtptime &= 0xffffffff;
- g_print ("rtptime now %lu then %u\n", ntprtptime, rtptime);
-
- if (ntprtptime > rtptime)
- ntptime -=
- gst_util_uint64_scale (ntprtptime - rtptime, jbuf->clock_rate,
- GST_SECOND);
- else
- ntptime +=
- gst_util_uint64_scale (rtptime - ntprtptime, jbuf->clock_rate,
- GST_SECOND);
-
- rtpsystime = gst_clock_adjust_unlocked (jbuf->media_clock, ntptime);
- g_print ("rtp %" GST_TIME_FORMAT " sys %" GST_TIME_FORMAT "\n",
- GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (systime));
- } else {
- g_print ("not synced yet\n");
- }
-
- /* do skew calculation by measuring the difference between rtptime and the
- * receive dts, this function will return the skew corrected rtptime. */
- item->pts = calculate_skew (jbuf, rtptime, dts);
-
append:
queue_do_insert (jbuf, list, (GList *) item);
/* ERRORS */
duplicate:
{
- GST_WARNING ("duplicate packet %d found", (gint) seqnum);
+ GST_DEBUG ("duplicate packet %d found", (gint) seqnum);
+ if (G_LIKELY (head))
+ *head = FALSE;
+ if (percent)
+ *percent = -1;
return FALSE;
}
}
/**
+ * rtp_jitter_buffer_alloc_item:
+ * @data: The data stored in this item
+ * @type: User specific item type
+ * @dts: Decoding Timestamp
+ * @pts: Presentation Timestamp
+ * @seqnum: Sequence number
+ * @count: Number of packet this item represent
+ * @rtptime: The RTP specific timestamp
+ * @free_data: A function to free @data (optional)
+ *
+ * Create an item that can then be stored in the jitter buffer.
+ *
+ * Returns: a newly allocated RTPJitterbufferItem
+ */
+static RTPJitterBufferItem *
+rtp_jitter_buffer_alloc_item (gpointer data, guint type, GstClockTime dts,
+ GstClockTime pts, guint seqnum, guint count, guint rtptime,
+ GDestroyNotify free_data)
+{
+ RTPJitterBufferItem *item;
+
+ item = g_slice_new (RTPJitterBufferItem);
+ item->data = data;
+ item->next = NULL;
+ item->prev = NULL;
+ item->type = type;
+ item->dts = dts;
+ item->pts = pts;
+ item->seqnum = seqnum;
+ item->count = count;
+ item->rtptime = rtptime;
+ item->free_data = free_data;
+
+ return item;
+}
+
+static inline RTPJitterBufferItem *
+alloc_event_item (GstEvent * event)
+{
+ return rtp_jitter_buffer_alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0,
+ -1, (GDestroyNotify) gst_mini_object_unref);
+}
+
+/**
+ * rtp_jitter_buffer_append_event:
+ * @jbuf: an #RTPJitterBuffer
+ * @event: an #GstEvent to insert
+
+ * Inserts @event into the packet queue of @jbuf.
+ *
+ * Returns: %TRUE if the event is at the head of the queue
+ */
+gboolean
+rtp_jitter_buffer_append_event (RTPJitterBuffer * jbuf, GstEvent * event)
+{
+ RTPJitterBufferItem *item = alloc_event_item (event);
+ gboolean head;
+ rtp_jitter_buffer_insert (jbuf, item, &head, NULL);
+ return head;
+}
+
+/**
+ * rtp_jitter_buffer_append_query:
+ * @jbuf: an #RTPJitterBuffer
+ * @query: an #GstQuery to insert
+
+ * Inserts @query into the packet queue of @jbuf.
+ *
+ * Returns: %TRUE if the query is at the head of the queue
+ */
+gboolean
+rtp_jitter_buffer_append_query (RTPJitterBuffer * jbuf, GstQuery * query)
+{
+ RTPJitterBufferItem *item =
+ rtp_jitter_buffer_alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1,
+ NULL);
+ gboolean head;
+ rtp_jitter_buffer_insert (jbuf, item, &head, NULL);
+ return head;
+}
+
+/**
+ * rtp_jitter_buffer_append_lost_event:
+ * @jbuf: an #RTPJitterBuffer
+ * @event: an #GstEvent to insert
+ * @seqnum: Sequence number
+ * @lost_packets: Number of lost packet this item represent
+
+ * Inserts @event into the packet queue of @jbuf.
+ *
+ * Returns: %TRUE if the event is at the head of the queue
+ */
+gboolean
+rtp_jitter_buffer_append_lost_event (RTPJitterBuffer * jbuf, GstEvent * event,
+ guint16 seqnum, guint lost_packets)
+{
+ RTPJitterBufferItem *item = rtp_jitter_buffer_alloc_item (event,
+ ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1,
+ (GDestroyNotify) gst_mini_object_unref);
+ gboolean head;
+
+ if (!rtp_jitter_buffer_insert (jbuf, item, &head, NULL)) {
+ /* Duplicate */
+ rtp_jitter_buffer_free_item (item);
+ head = FALSE;
+ }
+
+ return head;
+}
+
+/**
+ * rtp_jitter_buffer_append_buffer:
+ * @jbuf: an #RTPJitterBuffer
+ * @buf: an #GstBuffer to insert
+ * @seqnum: Sequence number
+ * @duplicate: TRUE when the packet inserted is a duplicate
+ * @percent: the buffering percent after insertion
+ *
+ * Inserts @buf into the packet queue of @jbuf.
+ *
+ * Returns: %TRUE if the buffer is at the head of the queue
+ */
+gboolean
+rtp_jitter_buffer_append_buffer (RTPJitterBuffer * jbuf, GstBuffer * buf,
+ GstClockTime dts, GstClockTime pts, guint16 seqnum, guint rtptime,
+ gboolean * duplicate, gint * percent)
+{
+ RTPJitterBufferItem *item = rtp_jitter_buffer_alloc_item (buf,
+ ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime,
+ (GDestroyNotify) gst_mini_object_unref);
+ gboolean head;
+ gboolean inserted;
+
+ inserted = rtp_jitter_buffer_insert (jbuf, item, &head, percent);
+ if (!inserted)
+ rtp_jitter_buffer_free_item (item);
+
+ if (duplicate)
+ *duplicate = !inserted;
+
+ return head;
+}
+
+/**
* rtp_jitter_buffer_pop:
* @jbuf: an #RTPJitterBuffer
* @percent: the buffering percent
*
* Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
- * have its timestamp adjusted with the incomming running_time and the detected
+ * have its timestamp adjusted with the incoming running_time and the detected
* clock skew.
*
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
g_return_val_if_fail (jbuf != NULL, NULL);
- queue = jbuf->packets;
+ queue = &jbuf->packets;
item = queue->head;
if (item) {
else if (percent)
*percent = -1;
+ /* let's clear the pointers so we can ensure we don't free items that are
+ * still in the jitterbuffer */
+ item->next = item->prev = NULL;
+
return (RTPJitterBufferItem *) item;
}
{
g_return_val_if_fail (jbuf != NULL, NULL);
- return (RTPJitterBufferItem *) jbuf->packets->head;
+ return (RTPJitterBufferItem *) jbuf->packets.head;
}
/**
* rtp_jitter_buffer_flush:
* @jbuf: an #RTPJitterBuffer
- * @free_func: function to free each item
+ * @free_func: function to free each item (optional)
* @user_data: user data passed to @free_func
*
* Flush all packets from the jitterbuffer.
GList *item;
g_return_if_fail (jbuf != NULL);
- g_return_if_fail (free_func != NULL);
- while ((item = g_queue_pop_head_link (jbuf->packets)))
+ if (free_func == NULL)
+ free_func = (GFunc) rtp_jitter_buffer_free_item;
+
+ while ((item = g_queue_pop_head_link (&jbuf->packets)))
free_func ((RTPJitterBufferItem *) item, user_data);
}
{
g_return_val_if_fail (jbuf != NULL, 0);
- return jbuf->packets->length;
+ return jbuf->packets.length;
}
/**
g_return_val_if_fail (jbuf != NULL, 0);
- high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
- low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
+ high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (&jbuf->packets);
+ low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (&jbuf->packets);
if (!high_buf || !low_buf || high_buf == low_buf)
return 0;
return result;
}
+
+/*
+ * rtp_jitter_buffer_get_seqnum_diff:
+ * @jbuf: an #RTPJitterBuffer
+ *
+ * Get the difference between the seqnum of first and last packet in the
+ * jitterbuffer.
+ *
+ * Returns: The difference expressed in seqnum.
+ */
+static guint16
+rtp_jitter_buffer_get_seqnum_diff (RTPJitterBuffer * jbuf)
+{
+ guint32 high_seqnum, low_seqnum;
+ RTPJitterBufferItem *high_buf, *low_buf;
+ guint16 result;
+
+ g_return_val_if_fail (jbuf != NULL, 0);
+
+ high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (&jbuf->packets);
+ low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (&jbuf->packets);
+
+ while (high_buf && high_buf->seqnum == -1)
+ high_buf = (RTPJitterBufferItem *) high_buf->prev;
+
+ while (low_buf && low_buf->seqnum == -1)
+ low_buf = (RTPJitterBufferItem *) low_buf->next;
+
+ if (!high_buf || !low_buf || high_buf == low_buf)
+ return 0;
+
+ high_seqnum = high_buf->seqnum;
+ low_seqnum = low_buf->seqnum;
+
+ /* it needs to work if ts wraps */
+ if (high_seqnum >= low_seqnum) {
+ result = (guint32) (high_seqnum - low_seqnum);
+ } else {
+ result = (guint32) (high_seqnum + G_MAXUINT16 + 1 - low_seqnum);
+ }
+ return result;
+}
+
/**
* rtp_jitter_buffer_get_sync:
* @jbuf: an #RTPJitterBuffer
if (last_rtptime)
*last_rtptime = jbuf->last_rtptime;
}
+
+/**
+ * rtp_jitter_buffer_can_fast_start:
+ * @jbuf: an #RTPJitterBuffer
+ * @num_packets: Number of consecutive packets needed
+ *
+ * Check if in the queue if there is enough packets with consecutive seqnum in
+ * order to start delivering them.
+ *
+ * Returns: %TRUE if the required number of consecutive packets was found.
+ */
+gboolean
+rtp_jitter_buffer_can_fast_start (RTPJitterBuffer * jbuf, gint num_packet)
+{
+ gboolean ret = TRUE;
+ RTPJitterBufferItem *last_item = NULL, *item;
+ gint i;
+
+ if (rtp_jitter_buffer_num_packets (jbuf) < num_packet)
+ return FALSE;
+
+ item = rtp_jitter_buffer_peek (jbuf);
+ for (i = 0; i < num_packet; i++) {
+ if (G_LIKELY (last_item)) {
+ guint16 expected_seqnum = last_item->seqnum + 1;
+
+ if (expected_seqnum != item->seqnum) {
+ ret = FALSE;
+ break;
+ }
+ }
+
+ last_item = item;
+ item = (RTPJitterBufferItem *) last_item->next;
+ }
+
+ return ret;
+}
+
+gboolean
+rtp_jitter_buffer_is_full (RTPJitterBuffer * jbuf)
+{
+ return rtp_jitter_buffer_get_seqnum_diff (jbuf) >= 32765 &&
+ rtp_jitter_buffer_num_packets (jbuf) > 10000;
+}
+
+
+/**
+ * rtp_jitter_buffer_free_item:
+ * @item: the item to be freed
+ *
+ * Free the jitter buffer item.
+ */
+void
+rtp_jitter_buffer_free_item (RTPJitterBufferItem * item)
+{
+ g_return_if_fail (item != NULL);
+ /* needs to be unlinked first */
+ g_return_if_fail (item->next == NULL);
+ g_return_if_fail (item->prev == NULL);
+
+ if (item->data && item->free_data)
+ item->free_data (item->data);
+ g_slice_free (RTPJitterBufferItem, item);
+}
projects / platform / upstream / gst-plugins-good.git / blobdiff