mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-09-30 07:42:32 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...

Browse source 
Original commit message from CVS:
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
qtdemux_parse_segments, qtdemux_parse_trak):
* gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
rtp_session_get_location, rtp_session_get_tool,
rtp_session_process_bye, session_report_blocks):
* gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
This commit is contained in:
Stefan Kost 2007-05-10 14:02:07 +00:00 committed by Tim-Philipp Müller
parent 091c2cfbc0
commit 15b54ec7e2
2 changed files with 24 additions and 19 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

24
gst/rtpmanager/rtpsession.c
View file

@ -364,7 +364,7 @@ rtp_session_get_bandwidth (RTPSession * sess)
* @bandwidth: the RTCP bandwidth
*
* Set the bandwidth that should be used for RTCP
* messages.
* messages.
*/
void
rtp_session_set_rtcp_bandwidth (RTPSession * sess, gdouble bandwidth)
@ -395,7 +395,7 @@ rtp_session_get_rtcp_bandwidth (RTPSession * sess)
* @sess: an #RTPSession
* @cname: a CNAME for the session
*
* Set the CNAME for the session.
* Set the CNAME for the session.
*/
void
rtp_session_set_cname (RTPSession * sess, const gchar * cname)
@ -427,7 +427,7 @@ rtp_session_get_cname (RTPSession * sess)
* @sess: an #RTPSession
* @name: a NAME for the session
*
* Set the NAME for the session.
* Set the NAME for the session.
*/
void
rtp_session_set_name (RTPSession * sess, const gchar * name)
@ -459,7 +459,7 @@ rtp_session_get_name (RTPSession * sess)
* @sess: an #RTPSession
* @email: an EMAIL for the session
*
* Set the EMAIL the session.
* Set the EMAIL the session.
*/
void
rtp_session_set_email (RTPSession * sess, const gchar * email)
@ -491,7 +491,7 @@ rtp_session_get_email (RTPSession * sess)
* @sess: an #RTPSession
* @phone: a PHONE for the session
*
* Set the PHONE the session.
* Set the PHONE the session.
*/
void
rtp_session_set_phone (RTPSession * sess, const gchar * phone)
@ -523,7 +523,7 @@ rtp_session_get_phone (RTPSession * sess)
* @sess: an #RTPSession
* @location: a LOCATION for the session
*
* Set the LOCATION the session.
* Set the LOCATION the session.
*/
void
rtp_session_set_location (RTPSession * sess, const gchar * location)
@ -555,7 +555,7 @@ rtp_session_get_location (RTPSession * sess)
* @sess: an #RTPSession
* @tool: a TOOL for the session
*
* Set the TOOL the session.
* Set the TOOL the session.
*/
void
rtp_session_set_tool (RTPSession * sess, const gchar * tool)
@ -587,7 +587,7 @@ rtp_session_get_tool (RTPSession * sess)
* @sess: an #RTPSession
* @note: a NOTE for the session
*
* Set the NOTE the session.
* Set the NOTE the session.
*/
void
rtp_session_set_note (RTPSession * sess, const gchar * note)
@ -1228,7 +1228,7 @@ rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
members = sess->stats.active_sources;
if (!sess->source->received_bye && members < pmembers) {
/* some members went away since the previous timeout estimate.
/* some members went away since the previous timeout estimate.
* Perform reverse reconsideration but only when we are not scheduling a
* BYE ourselves. */
if (arrival->time < sess->next_rtcp_check_time) {
@ -1612,7 +1612,8 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
GST_DEBUG ("ext_max %d, expected %d, received %d, base_seq %d",
GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
extended_max, expected, stats->packets_received, stats->base_seq);
lost = expected - stats->packets_received;
@ -1632,7 +1633,8 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
GST_DEBUG ("add RR for SSRC %08x", source->ssrc);
/* we scaled the jitter up for additional precision */
GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
extended_max, stats->jitter >> 4);
if (rtp_source_get_last_sr (source, &ntptime, NULL, NULL, NULL, &time)) {

19
gst/rtpmanager/rtpsource.c
View file

@ -347,7 +347,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
src->probation--;
src->stats.max_seq = seqnr;
if (src->probation == 0) {
GST_DEBUG ("probation done!", src->probation);
GST_DEBUG ("probation done!");
init_seq (src, seqnr);
} else {
GstBuffer *q;
@ -470,7 +470,8 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
/* push packet */
if (src->callbacks.push_rtp) {
GST_DEBUG ("pushing RTP packet %u", src->stats.packets_sent);
GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);
} else {
GST_DEBUG ("no callback installed");
@ -500,9 +501,10 @@ rtp_source_process_sr (RTPSource * src, guint64 ntptime, guint32 rtptime,
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %u, PC %u, OC %u",
src->ssrc, ntptime >> 32, ntptime & 0xffffffff, rtptime, packet_count,
octet_count);
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
(guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
packet_count, octet_count);
curridx = src->stats.curr_sr ^ 1;
curr = &src->stats.sr[curridx];
@ -543,9 +545,10 @@ rtp_source_process_rb (RTPSource * src, guint8 fractionlost, gint32 packetslost,
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got RB packet %d: SSRC %08x, FL %u"
", PL %u, HS %u, JITTER %u, LSR %08x, DLSR %08x", src->ssrc, fractionlost,
packetslost, exthighestseq, jitter, lsr, dlsr);
GST_DEBUG ("got RB packet: SSRC %08x, FL %" G_GUINT32_FORMAT ""
", PL %d, HS %" G_GUINT32_FORMAT ", JITTER %" G_GUINT32_FORMAT
", LSR %08x, DLSR %08x", src->ssrc, fractionlost, packetslost,
exthighestseq, jitter, lsr, dlsr);
curridx = src->stats.curr_rr ^ 1;
curr = &src->stats.rr[curridx];