*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#include <string.h>
#include <stdlib.h>
{RTP_JITTER_BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
{RTP_JITTER_BUFFER_MODE_BUFFER, "Do low/high watermark buffering",
"buffer"},
+ {RTP_JITTER_BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks",
+ "synced"},
{0, NULL, NULL},
};
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);
jbuf = RTP_JITTER_BUFFER_CAST (object);
- rtp_jitter_buffer_flush (jbuf);
- g_queue_free (jbuf->packets);
+ if (jbuf->media_clock_synced_id)
+ g_signal_handler_disconnect (jbuf->media_clock,
+ jbuf->media_clock_synced_id);
+ 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);
+
+ /* 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);
}
GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
}
+/**
+ * rtp_jitter_buffer_set_clock_rate:
+ * @jbuf: an #RTPJitterBuffer
+ * @clock_rate: the new clock rate
+ *
+ * Set the clock rate in the jitterbuffer.
+ */
+void
+rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
+{
+ if (jbuf->clock_rate != clock_rate) {
+ GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
+ G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
+ jbuf->clock_rate = clock_rate;
+ rtp_jitter_buffer_reset_skew (jbuf);
+ }
+}
+
+/**
+ * rtp_jitter_buffer_get_clock_rate:
+ * @jbuf: an #RTPJitterBuffer
+ *
+ * Get the currently configure clock rate in @jbuf.
+ *
+ * Returns: the current clock-rate
+ */
+guint32
+rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
+{
+ return jbuf->clock_rate;
+}
+
+static void
+media_clock_synced_cb (GstClock * clock, gboolean synced,
+ RTPJitterBuffer * jbuf)
+{
+ GstClockTime internal, external;
+
+ 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);
+ }
+ g_mutex_unlock (&jbuf->clock_lock);
+}
+
+/**
+ * rtp_jitter_buffer_set_media_clock:
+ * @jbuf: an #RTPJitterBuffer
+ * @clock: (transfer full): media #GstClock
+ * @clock_offset: RTP time at clock epoch or -1
+ *
+ * Sets the media clock for the media and the clock offset
+ *
+ */
+void
+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,
+ jbuf->media_clock_synced_id);
+ jbuf->media_clock_synced_id = 0;
+ gst_object_unref (jbuf->media_clock);
+ }
+ jbuf->media_clock = clock;
+ jbuf->media_clock_offset = clock_offset;
+
+ if (jbuf->pipeline_clock && jbuf->media_clock &&
+ jbuf->pipeline_clock != jbuf->media_clock) {
+ jbuf->media_clock_synced_id =
+ g_signal_connect (jbuf->media_clock, "synced",
+ G_CALLBACK (media_clock_synced_cb), jbuf);
+ if (gst_clock_is_synced (jbuf->media_clock)) {
+ GstClockTime internal, external;
+
+ 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_master (jbuf->media_clock, jbuf->pipeline_clock);
+ }
+ g_mutex_unlock (&jbuf->clock_lock);
+}
+
+/**
+ * rtp_jitter_buffer_set_pipeline_clock:
+ * @jbuf: an #RTPJitterBuffer
+ * @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 ? gst_object_ref (clock) : NULL;
+
+ if (jbuf->pipeline_clock && jbuf->media_clock &&
+ jbuf->pipeline_clock != jbuf->media_clock) {
+ if (gst_clock_is_synced (jbuf->media_clock)) {
+ GstClockTime internal, external;
+
+ 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_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;
+}
/**
* rtp_jitter_buffer_reset_skew:
jbuf->base_time = -1;
jbuf->base_rtptime = -1;
jbuf->base_extrtp = -1;
- jbuf->clock_rate = -1;
+ jbuf->media_clock_base_time = -1;
jbuf->ext_rtptime = -1;
jbuf->last_rtptime = -1;
jbuf->window_pos = 0;
jbuf->skew = 0;
jbuf->prev_send_diff = -1;
jbuf->prev_out_time = -1;
+ jbuf->need_resync = TRUE;
+
GST_DEBUG ("reset skew correction");
}
+/**
+ * rtp_jitter_buffer_disable_buffering:
+ * @jbuf: an #RTPJitterBuffer
+ * @disabled: the new state
+ *
+ * Enable or disable buffering on @jbuf.
+ */
+void
+rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
+{
+ jbuf->buffering_disabled = disabled;
+}
+
static void
rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
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;
jbuf->window_size = 0;
jbuf->skew = 0;
}
+ jbuf->need_resync = FALSE;
}
static guint64
get_buffer_level (RTPJitterBuffer * jbuf)
{
- GstBuffer *high_buf = NULL, *low_buf = NULL;
+ RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
guint64 level;
- GList *find;
- /* first first buffer with timestamp */
- find = g_queue_peek_head_link (jbuf->packets);
- while (find) {
- high_buf = find->data;
- if (GST_BUFFER_TIMESTAMP (high_buf) != -1)
+ /* first buffer with timestamp */
+ 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 = NULL;
- find = g_list_next (find);
+ high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
}
- find = g_queue_peek_tail_link (jbuf->packets);
- while (find) {
- low_buf = find->data;
- if (GST_BUFFER_TIMESTAMP (low_buf) != -1)
+ low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (&jbuf->packets);
+ while (low_buf) {
+ if (low_buf->dts != -1 || low_buf->pts != -1)
break;
- low_buf = NULL;
- find = g_list_previous (find);
+ low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
}
if (!high_buf || !low_buf || high_buf == low_buf) {
} else {
guint64 high_ts, low_ts;
- high_ts = GST_BUFFER_TIMESTAMP (high_buf);
- low_ts = GST_BUFFER_TIMESTAMP (low_buf);
+ high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
+ low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
if (high_ts > low_ts)
level = high_ts - low_ts;
level = get_buffer_level (jbuf);
GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
+ if (jbuf->buffering_disabled) {
+ GST_DEBUG ("buffering is disabled");
+ level = jbuf->high_level;
+ }
+
if (jbuf->buffering) {
post = TRUE;
- if (level > jbuf->high_level) {
+ if (level >= jbuf->high_level) {
GST_DEBUG ("buffering finished");
jbuf->buffering = FALSE;
}
* 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,
- guint32 clock_rate)
+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);
-
- gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
-
- /* keep track of the last extended rtptime */
- jbuf->last_rtptime = ext_rtptime;
-
- if (jbuf->clock_rate != clock_rate) {
- if (jbuf->clock_rate == -1) {
- GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
- G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
- } else {
- GST_WARNING ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
- G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
- }
- jbuf->base_time = -1;
- jbuf->base_rtptime = -1;
- jbuf->clock_rate = clock_rate;
- jbuf->prev_out_time = -1;
- jbuf->prev_send_diff = -1;
- }
-
- /* first time, lock on to time and gstrtptime */
- if (G_UNLIKELY (jbuf->base_time == -1)) {
- jbuf->base_time = time;
- jbuf->prev_out_time = -1;
- GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
- }
- if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
- jbuf->base_rtptime = gstrtptime;
- jbuf->base_extrtp = ext_rtptime;
- jbuf->prev_send_diff = -1;
- GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
- GST_TIME_ARGS (gstrtptime));
- }
-
- if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
- send_diff = gstrtptime - jbuf->base_rtptime;
- else if (time != -1) {
- /* elapsed time at sender, timestamps can go backwards and thus be smaller
- * than our base time, take a new base time in that case. */
- GST_WARNING ("backward timestamps at server, taking new base time");
- rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
- send_diff = 0;
- } else {
- GST_WARNING ("backward timestamps at server but no timestamps");
- send_diff = 0;
- /* at least try to get a new timestamp.. */
- jbuf->base_time = -1;
- }
-
- 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 */
* its timestamps. */
if (ABS (delta - jbuf->skew) > GST_SECOND) {
GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
- GST_TIME_ARGS (delta - jbuf->skew));
+ GST_TIME_ARGS (ABS (delta - jbuf->skew)));
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));
return out_time;
}
+static void
+queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
+{
+ GQueue *queue = &jbuf->packets;
+
+ /* It's more likely that the packet was inserted at the tail of the queue */
+ if (G_LIKELY (list)) {
+ item->prev = list;
+ item->next = list->next;
+ list->next = item;
+ } else {
+ item->prev = NULL;
+ item->next = queue->head;
+ queue->head = item;
+ }
+ if (item->next)
+ item->next->prev = item;
+ else
+ queue->tail = item;
+ 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
- * @buf: a buffer
- * @time: a running_time when this buffer was received in nanoseconds
- * @clock_rate: the clock-rate of the payload of @buf
- * @max_delay: the maximum lateness of @buf
- * @tail: TRUE when the tail element changed.
+ * @item: an #RTPJitterBufferItem to insert
+ * @head: TRUE when the head element changed.
+ * @percent: the buffering percent after insertion
*
- * Inserts @buf into the packet queue of @jbuf. The sequence number of the
+ * Inserts @item into the packet queue of @jbuf. The sequence number of the
* packet will be used to sort the packets. This function takes ownerhip of
* @buf when the function returns %TRUE.
- * @buf should have writable metadata when calling this function.
+ *
+ * When @head is %TRUE, the new packet was added at the head of the queue and
+ * will be available with the next call to rtp_jitter_buffer_pop() and
+ * rtp_jitter_buffer_peek().
*
* Returns: %FALSE if a packet with the same number already existed.
*/
-gboolean
-rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
- GstClockTime time, guint32 clock_rate, gboolean * tail, gint * percent)
+static gboolean
+rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
+ gboolean * head, gint * percent)
{
- GList *list;
- guint32 rtptime;
+ GList *list, *event = NULL;
guint16 seqnum;
- GstRTPBuffer rtp;
g_return_val_if_fail (jbuf != NULL, FALSE);
- g_return_val_if_fail (buf != NULL, FALSE);
+ g_return_val_if_fail (item != NULL, FALSE);
+
+ list = jbuf->packets.tail;
- gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+ /* no seqnum, simply append then */
+ if (item->seqnum == -1)
+ goto append;
- seqnum = gst_rtp_buffer_get_seq (&rtp);
+ seqnum = item->seqnum;
- /* loop the list to skip strictly smaller seqnum buffers */
- for (list = jbuf->packets->head; list; list = g_list_next (list)) {
+ /* loop the list to skip strictly larger seqnum buffers */
+ for (; list; list = g_list_previous (list)) {
guint16 qseq;
gint gap;
- GstRTPBuffer rtpb;
+ RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
+
+ if (qitem->seqnum == -1) {
+ /* keep a pointer to the first consecutive event if not already
+ * set. we will insert the packet after the event if we can't find
+ * a packet with lower sequence number before the event. */
+ if (event == NULL)
+ event = list;
+ continue;
+ }
- gst_rtp_buffer_map (GST_BUFFER_CAST (list->data), GST_MAP_READ, &rtpb);
- qseq = gst_rtp_buffer_get_seq (&rtpb);
- gst_rtp_buffer_unmap (&rtpb);
+ qseq = qitem->seqnum;
/* compare the new seqnum to the one in the buffer */
gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
/* seqnum > qseq, we can stop looking */
if (G_LIKELY (gap < 0))
break;
- }
- rtptime = gst_rtp_buffer_get_timestamp (&rtp);
- /* 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 (time) &&
- 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 * clock_rate ||
- ext_rtptime + 3 * 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);
- }
+ /* if we've found a packet with greater sequence number, cleanup the
+ * event pointer as the packet will be inserted before the event */
+ event = NULL;
}
- 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
- * interpollate 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)
- time = 0;
- else
- time = -1;
- break;
- case RTP_JITTER_BUFFER_MODE_SLAVE:
- default:
- break;
- }
- /* do skew calculation by measuring the difference between rtptime and the
- * receive time, this function will retimestamp @buf with the skew corrected
- * running time. */
- time = calculate_skew (jbuf, rtptime, time, clock_rate);
- GST_BUFFER_TIMESTAMP (buf) = time;
+ /* if event is set it means that packets before the event had smaller
+ * sequence number, so we will insert our packet after the event */
+ if (event)
+ list = event;
- /* It's more likely that the packet was inserted in the front of the buffer */
- if (G_LIKELY (list))
- g_queue_insert_before (jbuf->packets, list, buf);
- else
- g_queue_push_tail (jbuf->packets, buf);
+append:
+ queue_do_insert (jbuf, list, (GList *) item);
/* buffering mode, update buffer stats */
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
update_buffer_level (jbuf, percent);
- else
+ else if (percent)
*percent = -1;
- /* tail was changed when we did not find a previous packet, we set the return
+ /* head was changed when we did not find a previous packet, we set the return
* flag when requested. */
- if (G_LIKELY (tail))
- *tail = (list == NULL);
-
- gst_rtp_buffer_unmap (&rtp);
+ if (G_LIKELY (head))
+ *head = (list == NULL);
return TRUE;
/* ERRORS */
duplicate:
{
- gst_rtp_buffer_unmap (&rtp);
- 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.
*/
-GstBuffer *
+RTPJitterBufferItem *
rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
{
- GstBuffer *buf;
+ GList *item = NULL;
+ GQueue *queue;
g_return_val_if_fail (jbuf != NULL, NULL);
- buf = g_queue_pop_tail (jbuf->packets);
+ queue = &jbuf->packets;
+
+ item = queue->head;
+ if (item) {
+ queue->head = item->next;
+ if (queue->head)
+ queue->head->prev = NULL;
+ else
+ queue->tail = NULL;
+ queue->length--;
+ }
/* buffering mode, update buffer stats */
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
update_buffer_level (jbuf, percent);
- else
+ else if (percent)
*percent = -1;
- return buf;
+ /* 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;
}
/**
* rtp_jitter_buffer_peek:
* @jbuf: an #RTPJitterBuffer
*
- * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
- * with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
- * was inserted in the queue.
+ * Peek the oldest buffer from the packet queue of @jbuf.
+ *
+ * See rtp_jitter_buffer_insert() to check when an older packet was
+ * added.
*
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
*/
-GstBuffer *
+RTPJitterBufferItem *
rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
{
- GstBuffer *buf;
-
g_return_val_if_fail (jbuf != NULL, NULL);
- buf = g_queue_peek_tail (jbuf->packets);
-
- return buf;
+ return (RTPJitterBufferItem *) jbuf->packets.head;
}
/**
* rtp_jitter_buffer_flush:
* @jbuf: an #RTPJitterBuffer
+ * @free_func: function to free each item (optional)
+ * @user_data: user data passed to @free_func
*
* Flush all packets from the jitterbuffer.
*/
void
-rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf)
+rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
+ gpointer user_data)
{
- GstBuffer *buffer;
+ GList *item;
g_return_if_fail (jbuf != NULL);
- while ((buffer = g_queue_pop_head (jbuf->packets)))
- gst_buffer_unref (buffer);
+ 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);
}
/**
gboolean
rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
{
- return jbuf->buffering;
+ return jbuf->buffering && !jbuf->buffering_disabled;
}
/**
if (G_UNLIKELY (jbuf->high_level == 0))
return 100;
+ if (G_UNLIKELY (jbuf->buffering_disabled))
+ return 100;
+
level = get_buffer_level (jbuf);
percent = (level * 100 / jbuf->high_level);
percent = MIN (percent, 100);
{
g_return_val_if_fail (jbuf != NULL, 0);
- return jbuf->packets->length;
+ return jbuf->packets.length;
}
/**
rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
{
guint64 high_ts, low_ts;
- GstBuffer *high_buf, *low_buf;
+ RTPJitterBufferItem *high_buf, *low_buf;
guint32 result;
- GstRTPBuffer rtp;
g_return_val_if_fail (jbuf != NULL, 0);
- high_buf = g_queue_peek_head (jbuf->packets);
- low_buf = g_queue_peek_tail (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;
- gst_rtp_buffer_map (high_buf, GST_MAP_READ, &rtp);
- high_ts = gst_rtp_buffer_get_timestamp (&rtp);
- gst_rtp_buffer_unmap (&rtp);
- gst_rtp_buffer_map (low_buf, GST_MAP_READ, &rtp);
- low_ts = gst_rtp_buffer_get_timestamp (&rtp);
- gst_rtp_buffer_unmap (&rtp);
+ high_ts = high_buf->rtptime;
+ low_ts = low_buf->rtptime;
/* it needs to work if ts wraps */
if (high_ts >= low_ts) {
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