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-rw-r--r--ChangeLog21
-rw-r--r--gst/qtdemux/qtdemux.c24
-rw-r--r--gst/rtpmanager/rtpsession.c24
-rw-r--r--gst/rtpmanager/rtpsource.c19
-rw-r--r--gst/switch/Makefile.am2
5 files changed, 60 insertions, 30 deletions
diff --git a/ChangeLog b/ChangeLog
index fb03cc33..a5a9594d 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,24 @@
2007-05-10 Stefan Kost <ensonic@users.sf.net>
+ * 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>).
+
+2007-05-10 Stefan Kost <ensonic@users.sf.net>
+
* gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
async_jitter_queue_set_low_threshold,
@@ -11,7 +30,7 @@
async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
async_jitter_queue_set_flushing_unlocked,
async_jitter_queue_unset_flushing_unlocked):
- Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
+ Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>).
2007-05-10 Stefan Kost <ensonic@users.sf.net>
diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c
index 66bc04ea..ec438d5c 100644
--- a/gst/qtdemux/qtdemux.c
+++ b/gst/qtdemux/qtdemux.c
@@ -585,7 +585,7 @@ gst_qtdemux_move_stream (GstQTDemux * qtdemux, QtDemuxStream * str,
* streaming from the desired position.
*
* Keyframe seeking is a little more complicated when dealing with
- * segments. Ideally we want to move to the previous keyframe in
+ * segments. Ideally we want to move to the previous keyframe in
* the segment but there might not be a keyframe in the segment. In
* fact, none of the segments could contain a keyframe. We take a
* practical approach: seek to the previous keyframe in the segment,
@@ -1024,7 +1024,7 @@ beach:
* @offset is an absolute global position over all the segments.
*
* This will push out a NEWSEGMENT event with the right values and
- * position the stream index to the first decodable sample before
+ * position the stream index to the first decodable sample before
* @offset.
*/
static gboolean
@@ -1107,7 +1107,7 @@ gst_qtdemux_activate_segment (GstQTDemux * qtdemux, QtDemuxStream * stream,
}
/* prepare to get the current sample of @stream, getting essential values.
- *
+ *
* This function will also prepare and send the segment when needed.
*
* Return FALSE if the stream is EOS.
@@ -1142,6 +1142,9 @@ gst_qtdemux_prepare_current_sample (GstQTDemux * qtdemux,
if (stream->segment_index != seg_idx)
gst_qtdemux_activate_segment (qtdemux, stream, seg_idx, time_position);
+ GST_LOG_OBJECT (qtdemux, "segment active, index = %lu of %lu",
+ stream->sample_index, stream->n_samples);
+
if (stream->sample_index >= stream->n_samples)
goto eos;
@@ -1248,6 +1251,7 @@ gst_qtdemux_combine_flows (GstQTDemux * demux, QtDemuxStream * stream,
/* if we get here, all other pads were unlinked and we return
* NOT_LINKED then */
done:
+ GST_LOG_OBJECT (demux, "combined flow return: %s", gst_flow_get_name (ret));
return ret;
}
@@ -1350,8 +1354,9 @@ gst_qtdemux_loop_state_movie (GstQTDemux * qtdemux)
GST_LOG_OBJECT (qtdemux,
"Pushing buffer with time %" GST_TIME_FORMAT ", duration %"
- GST_TIME_FORMAT " on pad %p", GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (duration), stream->pad);
+ GST_TIME_FORMAT " on pad %s",
+ GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
+ GST_PAD_NAME (stream->pad));
ret = gst_pad_push (stream->pad, buf);
} else {
ret = GST_FLOW_OK;
@@ -1464,7 +1469,7 @@ pause:
/*
* next_entry_size
- *
+ *
* Returns the size of the first entry at the current offset.
* If -1, there are none (which means EOS or empty file).
*/
@@ -2613,7 +2618,7 @@ done:
/* parse the traks.
* With each track we associate a new QtDemuxStream that contains all the info
- * about the trak.
+ * about the trak.
* traks that do not decode to something (like strm traks) will not have a pad.
*/
static gboolean
@@ -2667,7 +2672,7 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak)
stream->duration = QT_UINT32 ((guint8 *) mdhd->data + 24);
}
- GST_LOG_OBJECT (qtdemux, "track timescale: %" G_GUINT64_FORMAT,
+ GST_LOG_OBJECT (qtdemux, "track timescale: %" G_GUINT32_FORMAT,
stream->timescale);
GST_LOG_OBJECT (qtdemux, "track duration: %" G_GUINT64_FORMAT,
stream->duration);
@@ -2685,7 +2690,8 @@ qtdemux_parse_trak (GstQTDemux * qtdemux, GNode * trak)
* identify those yet, except for just looking at their duration. */
if (tdur1 != 0 && (tdur2 * 10 / tdur1) < 2) {
GST_WARNING_OBJECT (qtdemux,
- "Track shorter than 20%% (%d/%d vs. %d/%d) of the stream "
+ "Track shorter than 20%% (%" G_GUINT64_FORMAT "/%" G_GUINT32_FORMAT
+ " vs. %" G_GUINT32_FORMAT "/%" G_GUINT32_FORMAT ") of the stream "
"found, assuming preview image or something; skipping track",
stream->duration, stream->timescale, qtdemux->duration,
qtdemux->timescale);
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 9f5d58d8..74907912 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -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)) {
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 7af74671..8007d54b 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -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];
diff --git a/gst/switch/Makefile.am b/gst/switch/Makefile.am
index fcac882e..b5a55038 100644
--- a/gst/switch/Makefile.am
+++ b/gst/switch/Makefile.am
@@ -4,6 +4,6 @@ plugin_LTLIBRARIES = libgstswitch.la
libgstswitch_la_SOURCES = gstswitch.c
libgstswitch_la_CFLAGS = $(GST_CFLAGS)
libgstswitch_la_LIBADD =
-libgstswitch_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstswitch_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS)
noinst_HEADERS = gstswitch.h
generated by cgit v1.2.1 (git 2.18.0) at 2024-07-05 18:53:27 +0000