/* GStreamer * Copyright (C) <2007> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #include #include #include #include #include "rtpjitterbuffer.h" GST_DEBUG_CATEGORY_STATIC (rtp_jitter_buffer_debug); #define GST_CAT_DEFAULT rtp_jitter_buffer_debug #define MAX_WINDOW RTP_JITTER_BUFFER_MAX_WINDOW #define MAX_TIME (2 * GST_SECOND) /* signals and args */ enum { LAST_SIGNAL }; enum { PROP_0 }; /* GObject vmethods */ static void rtp_jitter_buffer_finalize (GObject * object); GType rtp_jitter_buffer_mode_get_type (void) { static GType jitter_buffer_mode_type = 0; static const GEnumValue jitter_buffer_modes[] = { {RTP_JITTER_BUFFER_MODE_NONE, "Only use RTP timestamps", "none"}, {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}, }; if (!jitter_buffer_mode_type) { jitter_buffer_mode_type = g_enum_register_static ("RTPJitterBufferMode", jitter_buffer_modes); } return jitter_buffer_mode_type; } /* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */ G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT); static void rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass) { GObjectClass *gobject_class; gobject_class = (GObjectClass *) klass; gobject_class->finalize = rtp_jitter_buffer_finalize; GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer"); } static void rtp_jitter_buffer_init (RTPJitterBuffer * jbuf) { g_mutex_init (&jbuf->clock_lock); jbuf->packets = g_queue_new (); jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE; rtp_jitter_buffer_reset_skew (jbuf); } static void rtp_jitter_buffer_finalize (GObject * object) { RTPJitterBuffer *jbuf; jbuf = RTP_JITTER_BUFFER_CAST (object); if (jbuf->media_clock_synced_id) g_signal_handler_disconnect (jbuf->media_clock, jbuf->media_clock_synced_id); if (jbuf->media_clock) gst_object_unref (jbuf->media_clock); if (jbuf->pipeline_clock) gst_object_unref (jbuf->pipeline_clock); g_queue_free (jbuf->packets); g_mutex_clear (&jbuf->clock_lock); G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object); } /** * rtp_jitter_buffer_new: * * Create an #RTPJitterBuffer. * * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage. */ RTPJitterBuffer * rtp_jitter_buffer_new (void) { RTPJitterBuffer *jbuf; jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL); return jbuf; } /** * rtp_jitter_buffer_get_mode: * @jbuf: an #RTPJitterBuffer * * Get the current jitterbuffer mode. * * Returns: the current jitterbuffer mode. */ RTPJitterBufferMode rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf) { return jbuf->mode; } /** * rtp_jitter_buffer_set_mode: * @jbuf: an #RTPJitterBuffer * @mode: a #RTPJitterBufferMode * * Set the buffering and clock slaving algorithm used in the @jbuf. */ void rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode) { jbuf->mode = mode; } GstClockTime rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf) { return jbuf->delay; } void rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay) { jbuf->delay = delay; jbuf->low_level = (delay * 15) / 100; /* the high level is at 90% in order to release packets before we fill up the * buffer up to the latency */ jbuf->high_level = (delay * 90) / 100; GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay), 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: an #RTPJitterBuffer * * Reset the skew calculations in @jbuf. */ void rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf) { 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->window_filling = TRUE; jbuf->window_min = 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->prev_send_diff = -1; if (reset_skew) { jbuf->window_filling = TRUE; jbuf->window_pos = 0; jbuf->window_min = 0; jbuf->window_size = 0; jbuf->skew = 0; } jbuf->need_resync = FALSE; } static guint64 get_buffer_level (RTPJitterBuffer * jbuf) { RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL; guint64 level; /* 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 = (RTPJitterBufferItem *) g_list_previous (high_buf); } 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 = (RTPJitterBufferItem *) g_list_next (low_buf); } if (!high_buf || !low_buf || high_buf == low_buf) { level = 0; } else { guint64 high_ts, low_ts; 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; else level = 0; GST_LOG_OBJECT (jbuf, "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %" G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts), level); } return level; } static void update_buffer_level (RTPJitterBuffer * jbuf, gint * percent) { gboolean post = FALSE; guint64 level; 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) { GST_DEBUG ("buffering finished"); jbuf->buffering = FALSE; } } else { if (level < jbuf->low_level) { GST_DEBUG ("buffering started"); jbuf->buffering = TRUE; post = TRUE; } } if (post) { gint perc; if (jbuf->buffering && (jbuf->high_level != 0)) { perc = (level * 100 / jbuf->high_level); perc = MIN (perc, 100); } else { perc = 100; } if (percent) *percent = perc; GST_DEBUG ("buffering %d", perc); } } /* For the clock skew we use a windowed low point averaging algorithm as can be * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation * over Network Delays": * http://www.grame.fr/Ressources/pub/TR-050601.pdf * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546 * * The idea is that the jitter is composed of: * * J = N + n * * N : a constant network delay. * n : random added noise. The noise is concentrated around 0 * * In the receiver we can track the elapsed time at the sender with: * * send_diff(i) = (Tsi - Ts0); * * Tsi : The time at the sender at packet i * Ts0 : The time at the sender at the first packet * * This is the difference between the RTP timestamp in the first received packet * and the current packet. * * At the receiver we have to deal with the jitter introduced by the network. * * recv_diff(i) = (Tri - Tr0) * * Tri : The time at the receiver at packet i * Tr0 : The time at the receiver at the first packet * * Both of these values contain a jitter Ji, a jitter for packet i, so we can * write: * * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0)) * * 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: * * recv_diff(i) = (Cri + ni) - (Cr0 + n0)) * * The drift is now expressed as: * * Drift(i) = recv_diff(i) - send_diff(i); * * We now keep the W latest values of Drift and find the minimum (this is the * one with the lowest network jitter and thus the one which is least affected * by it). We average this lowest value to smooth out the resulting network skew. * * 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. * * We use a 2 second window or up to 512 data points, which is statistically big * enough to catch spikes (FIXME, detect spikes). * We also use a rather large weighting factor (125) to smoothly adapt. During * startup, when filling the window, we use a parabolic weighting factor, the * more the window is filled, the faster we move to the detected possible skew. * * Returns: @time adjusted with the clock skew. */ static GstClockTime calculate_skew (RTPJitterBuffer * jbuf, guint64 ext_rtptime, GstClockTime gstrtptime, GstClockTime time) { guint64 send_diff, recv_diff; gint64 delta; gint64 old; gint pos, i; GstClockTime out_time; guint64 slope; /* 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) goto no_skew; /* elapsed time at receiver, includes the jitter */ recv_diff = time - jbuf->base_time; /* measure the diff */ delta = ((gint64) recv_diff) - ((gint64) send_diff); /* measure the slope, this gives a rought estimate between the sender speed * and the receiver speed. This should be approximately 8, higher values * indicate a burst (especially when the connection starts) */ if (recv_diff > 0) slope = (send_diff * 8) / recv_diff; else slope = 8; GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %" GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope); /* if the difference between the sender timeline and the receiver timeline * changed too quickly we have to resync because the server likely restarted * its timestamps. */ if (ABS (delta - jbuf->skew) > GST_SECOND) { GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew", GST_TIME_ARGS (ABS (delta - jbuf->skew))); rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE); send_diff = 0; delta = 0; } pos = jbuf->window_pos; if (G_UNLIKELY (jbuf->window_filling)) { /* we are filling the window */ GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta); jbuf->window[pos++] = delta; /* calc the min delta we observed */ if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min)) jbuf->window_min = delta; if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) { jbuf->window_size = pos; /* window filled */ GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min); /* the skew is now the min */ jbuf->skew = jbuf->window_min; jbuf->window_filling = FALSE; } else { gint perc_time, perc_window, perc; /* figure out how much we filled the window, this depends on the amount of * time we have or the max number of points we keep. */ perc_time = send_diff * 100 / MAX_TIME; perc_window = pos * 100 / MAX_WINDOW; perc = MAX (perc_time, perc_window); /* make a parabolic function, the closer we get to the MAX, the more value * we give to the scaling factor of the new value */ perc = perc * perc; /* quickly go to the min value when we are filling up, slowly when we are * just starting because we're not sure it's a good value yet. */ jbuf->skew = (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000; jbuf->window_size = pos + 1; } } else { /* pick old value and store new value. We keep the previous value in order * to quickly check if the min of the window changed */ old = jbuf->window[pos]; jbuf->window[pos++] = delta; if (G_UNLIKELY (delta <= jbuf->window_min)) { /* if the new value we inserted is smaller or equal to the current min, * it becomes the new min */ jbuf->window_min = delta; } else if (G_UNLIKELY (old == jbuf->window_min)) { gint64 min = G_MAXINT64; /* if we removed the old min, we have to find a new min */ for (i = 0; i < jbuf->window_size; i++) { /* we found another value equal to the old min, we can stop searching now */ if (jbuf->window[i] == old) { min = old; break; } if (jbuf->window[i] < min) min = jbuf->window[i]; } jbuf->window_min = min; } /* average the min values */ jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125; GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT, delta, jbuf->window_min); } /* wrap around in the window */ if (G_UNLIKELY (pos >= jbuf->window_size)) pos = 0; jbuf->window_pos = pos; no_skew: /* the output time is defined as the base timestamp plus the RTP time * adjusted for the clock skew .*/ if (jbuf->base_time != -1) { out_time = jbuf->base_time + send_diff; /* skew can be negative and we don't want to make invalid timestamps */ if (jbuf->skew < 0 && out_time < -jbuf->skew) { out_time = 0; } else { out_time += jbuf->skew; } } else out_time = -1; 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++; } /** * rtp_jitter_buffer_insert: * @jbuf: an #RTPJitterBuffer * @item: an #RTPJitterBufferItem to insert * @head: TRUE when the head element changed. * @percent: the buffering percent after insertion * @base_time: base time of the pipeline * * 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. * * 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, RTPJitterBufferItem * item, gboolean * head, gint * percent, GstClockTime base_time) { GList *list, *event = NULL; guint32 rtptime; guint64 ext_rtptime; guint16 seqnum; GstClockTime gstrtptime, dts; GstClock *media_clock, *pipeline_clock; guint64 media_clock_offset; gboolean rfc7273_mode; g_return_val_if_fail (jbuf != NULL, FALSE); g_return_val_if_fail (item != NULL, FALSE); list = jbuf->packets->tail; /* no seqnum, simply append then */ if (item->seqnum == -1) goto append; seqnum = item->seqnum; /* loop the list to skip strictly larger seqnum buffers */ for (; list; list = g_list_previous (list)) { guint16 qseq; gint gap; 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; } qseq = qitem->seqnum; /* compare the new seqnum to the one in the buffer */ gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq); /* we hit a packet with the same seqnum, notify a duplicate */ if (G_UNLIKELY (gap == 0)) goto duplicate; /* seqnum > qseq, we can stop looking */ if (G_LIKELY (gap < 0)) break; /* 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; } /* 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; 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); } } /* 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) { item->pts = jbuf->prev_out_time; goto append; } /* 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) { /* 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; } } 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)) { 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) item->pts = rtpsystime - base_time; else item->pts = 0; GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT, GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (item->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, jbuf->clock_rate, GST_SECOND); else ntptime += gst_util_uint64_scale (rtptime_tmp - ntprtptime, jbuf->clock_rate, GST_SECOND); 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) item->pts = rtpsystime - base_time; else item->pts = 0; GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT, GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (item->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. */ item->pts = calculate_skew (jbuf, ext_rtptime, gstrtptime, dts); } /* 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 (item->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 && item->pts < jbuf->prev_out_time) || /* if the server timestamps went backwards and the out_time forwards */ (gstrtptime - jbuf->base_rtptime < jbuf->prev_send_diff && item->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"); item->pts = jbuf->prev_out_time; } } if (dts != -1 && item->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", GST_TIME_ARGS (item->pts), jbuf->delay, GST_TIME_ARGS (dts)); rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, TRUE); item->pts = dts; } jbuf->prev_out_time = item->pts; jbuf->prev_send_diff = gstrtptime - jbuf->base_rtptime; if (media_clock) gst_object_unref (media_clock); if (pipeline_clock) gst_object_unref (pipeline_clock); 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 if (percent) *percent = -1; /* head was changed when we did not find a previous packet, we set the return * flag when requested. */ if (G_LIKELY (head)) *head = (list == NULL); return TRUE; /* ERRORS */ duplicate: { GST_DEBUG ("duplicate packet %d found", (gint) seqnum); return FALSE; } } /** * 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 * clock skew. * * Returns: a #GstBuffer or %NULL when there was no packet in the queue. */ RTPJitterBufferItem * rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent) { GList *item = NULL; GQueue *queue; g_return_val_if_fail (jbuf != NULL, NULL); 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 if (percent) *percent = -1; return (RTPJitterBufferItem *) item; } /** * rtp_jitter_buffer_peek: * @jbuf: an #RTPJitterBuffer * * 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. */ RTPJitterBufferItem * rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf) { g_return_val_if_fail (jbuf != NULL, NULL); return (RTPJitterBufferItem *) jbuf->packets->head; } /** * rtp_jitter_buffer_flush: * @jbuf: an #RTPJitterBuffer * @free_func: function to free each item * @user_data: user data passed to @free_func * * Flush all packets from the jitterbuffer. */ void rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func, gpointer user_data) { GList *item; g_return_if_fail (jbuf != NULL); g_return_if_fail (free_func != NULL); while ((item = g_queue_pop_head_link (jbuf->packets))) free_func ((RTPJitterBufferItem *) item, user_data); } /** * rtp_jitter_buffer_is_buffering: * @jbuf: an #RTPJitterBuffer * * Check if @jbuf is buffering currently. Users of the jitterbuffer should not * pop packets while in buffering mode. * * Returns: the buffering state of @jbuf */ gboolean rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf) { return jbuf->buffering && !jbuf->buffering_disabled; } /** * rtp_jitter_buffer_set_buffering: * @jbuf: an #RTPJitterBuffer * @buffering: the new buffering state * * Forces @jbuf to go into the buffering state. */ void rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering) { jbuf->buffering = buffering; } /** * rtp_jitter_buffer_get_percent: * @jbuf: an #RTPJitterBuffer * * Get the buffering percent of the jitterbuffer. * * Returns: the buffering percent */ gint rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf) { gint percent; guint64 level; 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); return percent; } /** * rtp_jitter_buffer_num_packets: * @jbuf: an #RTPJitterBuffer * * Get the number of packets currently in "jbuf. * * Returns: The number of packets in @jbuf. */ guint rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf) { g_return_val_if_fail (jbuf != NULL, 0); return jbuf->packets->length; } /** * rtp_jitter_buffer_get_ts_diff: * @jbuf: an #RTPJitterBuffer * * Get the difference between the timestamps of first and last packet in the * jitterbuffer. * * Returns: The difference expressed in the timestamp units of the packets. */ guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf) { guint64 high_ts, low_ts; RTPJitterBufferItem *high_buf, *low_buf; guint32 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); if (!high_buf || !low_buf || high_buf == low_buf) return 0; high_ts = high_buf->rtptime; low_ts = low_buf->rtptime; /* it needs to work if ts wraps */ if (high_ts >= low_ts) { result = (guint32) (high_ts - low_ts); } else { result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts); } return result; } /** * rtp_jitter_buffer_get_sync: * @jbuf: an #RTPJitterBuffer * @rtptime: result RTP time * @timestamp: result GStreamer timestamp * @clock_rate: clock-rate of @rtptime * @last_rtptime: last seen rtptime. * * Calculates the relation between the RTP timestamp and the GStreamer timestamp * used for constructing timestamps. * * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate, * the GStreamer timestamp is currently @timestamp. * * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in * @last_rtptime. */ void rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime, guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime) { if (rtptime) *rtptime = jbuf->base_extrtp; if (timestamp) *timestamp = jbuf->base_time + jbuf->skew; if (clock_rate) *clock_rate = jbuf->clock_rate; if (last_rtptime) *last_rtptime = jbuf->last_rtptime; }