/*
* Farsight Voice+Video library
*
- * Copyright 2007 Collabora Ltd,
+ * Copyright 2007 Collabora Ltd,
* Copyright 2007 Nokia Corporation
* @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
* Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
* from a network source. It will also wait for missing packets up to a
* configurable time limit using the #GstRtpJitterBuffer:latency property.
* Packets arriving too late are considered to be lost packets.
- *
+ *
* This element acts as a live element and so adds #GstRtpJitterBuffer:latency
* to the pipeline.
- *
+ *
* The element needs the clock-rate of the RTP payload in order to estimate the
* delay. This information is obtained either from the caps on the sink pad or,
* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
* To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
- *
+ *
* This element will automatically be used inside gstrtpbin.
- *
+ *
* <refsect2>
* <title>Example pipelines</title>
* |[
/**
* GstRtpJitterBuffer::latency:
- *
+ *
* The maximum latency of the jitterbuffer. Packets will be kept in the buffer
* for at most this time.
*/
G_PARAM_READWRITE));
/**
* GstRtpJitterBuffer::drop-on-latency:
- *
- * Drop oldest buffers when the queue is completely filled.
+ *
+ * Drop oldest buffers when the queue is completely filled.
*/
g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
g_param_spec_boolean ("drop-on-latency",
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE));
/**
* GstRtpJitterBuffer::ts-offset:
- *
+ *
* Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
* This is mainly used to ensure interstream synchronisation.
*/
/**
* GstRtpJitterBuffer::do-lost:
- *
+ *
* Send out a GstRTPPacketLost event downstream when a packet is considered
* lost.
*/
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
/* this unblocks any waiting pops on the src pad task */
JBUF_SIGNAL (priv);
- /* unlock clock, we just unschedule, the entry will be released by the
+ /* unlock clock, we just unschedule, the entry will be released by the
* locking streaming thread. */
if (priv->clock_id) {
gst_clock_id_unschedule (priv->clock_id);
* FALSE if a packet with the same seqnum was already in the queue, meaning we
* have a duplicate. */
if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
- priv->clock_rate, &tail)))
+ priv->clock_rate, (priv->latency_ms * GST_MSECOND), &tail)))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
*
* Both the window and the weighting used for averaging influence the accuracy
* of the drift estimation. Finding the correct parameters turns out to be a
- * compromise between accuracy and inertia.
+ * compromise between accuracy and inertia.
*
* We use a 2 second window or up to 512 data points, which is statistically big
* enough to catch spikes (FIXME, detect spikes).
*/
static GstClockTime
calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
- guint32 clock_rate)
+ guint32 clock_rate, GstClockTime max_delay)
{
guint64 ext_rtptime;
guint64 send_diff, recv_diff;
* changed too quickly we have to resync because the server likely restarted
* its timestamps. */
if (ABS (delta - jbuf->skew) > GST_SECOND) {
- GST_WARNING ("delta %" GST_TIME_FORMAT " too big, reset skew",
+ GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
GST_TIME_ARGS (delta - jbuf->skew));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
send_diff = 0;
out_time = jbuf->prev_out_time;
}
}
+
+ if (out_time + max_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),
+ max_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;
* @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.
*
* Inserts @buf into the packet queue of @jbuf. The sequence number of the
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
- GstClockTime time, guint32 clock_rate, gboolean * tail)
+ GstClockTime time, guint32 clock_rate, GstClockTime max_delay,
+ gboolean * tail)
{
GList *list;
guint32 rtptime;
* receive time, this function will retimestamp @buf with the skew corrected
* running time. */
rtptime = gst_rtp_buffer_get_timestamp (buf);
- time = calculate_skew (jbuf, rtptime, time, clock_rate);
+ time = calculate_skew (jbuf, rtptime, time, clock_rate, max_delay);
GST_BUFFER_TIMESTAMP (buf) = time;
/* It's more likely that the packet was inserted in the front of the buffer */
void rtp_jitter_buffer_reset_skew (RTPJitterBuffer *jbuf);
gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf, GstBuffer *buf,
- GstClockTime time,
- guint32 clock_rate,
- gboolean *tail);
+ GstClockTime time,
+ guint32 clock_rate,
+ GstClockTime max_delay,
+ gboolean *tail);
GstBuffer * rtp_jitter_buffer_peek (RTPJitterBuffer *jbuf);
GstBuffer * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf);
void rtp_jitter_buffer_get_sync (RTPJitterBuffer *jbuf, guint64 *rtptime,
guint64 *timestamp, guint32 *clock_rate,
- guint64 *last_rtptime);
-
+ guint64 *last_rtptime);
#endif /* __RTP_JITTER_BUFFER_H__ */
projects / platform / upstream / gst-plugins-good.git / commitdiff