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webrtc_m130/webrtc/modules/rtp_rtcp/source
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skvlad 1c392cc5cf Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change. 
The change made in https://codereview.webrtc.org/1757683002 introduced an extra call to RTCPSender::SetRTCPStatus after the video receive stream is created. The SetRTCPStatus call results in no state change, as the RTCP sender is already enabled, however, it reschedules the next RTCP packet to be RTCP_INTERVAL_VIDEO_MS/2 (500) ms in the future.
Before the change, the next packet time was only set by the previous call to RTCPSender::SetSSRC, which placed it 100 ms in the future. The change, therefore, changed the timing of multiple performance tests - as it now takes a different length of time to ramp up to the same bandwidth.

BUG=chromium:597332

Review URL: https://codereview.webrtc.org/1826093004

Cr-Commit-Position: refs/heads/master@{#12203}
2016-04-01 21:46:54 +00:00
..
mock
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtcp_packet
[rtcp] Sdes::Parse updated not to use RTCPUtility
2016-03-30 20:34:38 +00:00
bitrate.cc
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
bitrate.h
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
2015-02-26 14:43:50 +00:00
byte_io_unittest.cc
Fix ubsan warning in byteio_unittest
2016-02-26 13:13:51 +00:00
byte_io.h
Fix a bug in webrtc::ByteReader
2016-01-25 07:47:59 +00:00
CPPLINT.cfg
[rtp_rtcp] Lint errors cleaned from rtp_utility
2015-12-28 18:18:52 +00:00
dtmf_queue.cc
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00
dtmf_queue.h
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
fec_private_tables_bursty.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
fec_private_tables_random.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
fec_receiver_impl.cc
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
fec_receiver_impl.h
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
fec_receiver_unittest.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
fec_test_helper.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
fec_test_helper.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
forward_error_correction_internal.cc
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
forward_error_correction_internal.h
Add include guards to forward_error_correction_internal.h
2013-12-24 05:06:12 +00:00
forward_error_correction.cc
Use pointer to generated FEC packet.
2016-01-07 14:33:57 +00:00
forward_error_correction.h
Replace RefCountImpl with rtc::RefCountedObject.
2016-03-21 15:44:41 +00:00
h264_bitstream_parser_unittest.cc
H264 bitstream parser.
2015-09-24 13:07:04 +00:00
h264_bitstream_parser.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
h264_bitstream_parser.h
lint build/include errors fixed in rtp_rtcp module
2015-12-10 13:05:35 +00:00
h264_sps_parser_unittest.cc
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
h264_sps_parser.cc
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
h264_sps_parser.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
nack_rtx_unittest.cc
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
packet_loss_stats_unittest.cc
Add statistics gathering for packet loss.
2015-07-11 01:10:08 +00:00
packet_loss_stats.cc
Add RTC_ prefix to (D)CHECKs and related macros.
2015-09-17 07:24:51 +00:00
packet_loss_stats.h
Add statistics gathering for packet loss.
2015-07-11 01:10:08 +00:00
producer_fec_unittest.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
producer_fec.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
producer_fec.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
receive_statistics_impl.cc
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
2016-02-25 12:50:09 +00:00
receive_statistics_impl.h
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
2016-02-25 12:50:09 +00:00
receive_statistics_unittest.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
remote_ntp_time_estimator_unittest.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
remote_ntp_time_estimator.cc
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +00:00
rtcp_format_remb_unittest.cc
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
2016-01-21 13:42:10 +00:00
rtcp_packet_unittest.cc
[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
2016-01-18 10:43:38 +00:00
rtcp_packet.cc
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_packet.h
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_receiver_help.cc
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
rtcp_receiver_help.h
lint build/include errors fixed in rtp_rtcp module
2015-12-10 13:05:35 +00:00
rtcp_receiver_unittest.cc
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +00:00
rtcp_receiver.cc
Enabling rtcp-rsize negotiation and fixing some issues with it.
2016-03-18 22:02:13 +00:00
rtcp_receiver.h
Enabling rtcp-rsize negotiation and fixing some issues with it.
2016-03-18 22:02:13 +00:00
rtcp_sender_unittest.cc
RTT intermediate calculation use ntp time instead of milliseconds.
2016-02-22 17:59:47 +00:00
rtcp_sender.cc
Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change.
2016-04-01 21:46:54 +00:00
rtcp_sender.h
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtcp_utility_unittest.cc
Add a ParseHeader method to RtcpPacket, for parsing common RTCP header.
2015-09-14 19:50:49 +00:00
rtcp_utility.cc
Fixed busy loop in case of partially malformed rtcp packet
2016-03-10 21:13:57 +00:00
rtcp_utility.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
rtp_fec_unittest.cc
[rtp_rtcp] lint errors about rand() usage fixed.
2015-12-15 08:30:12 +00:00
rtp_format_h264_unittest.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_h264.cc
Add SEI to the list of packets counted as keyframe.
2016-02-03 11:43:50 +00:00
rtp_format_h264.h
Consolidate constructormagic macros with Chromium version and remove Chromium override.
2015-09-16 13:33:25 +00:00
rtp_format_video_generic.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_video_generic.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
rtp_format_vp8_test_helper.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_format_vp8_test_helper.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_vp8_unittest.cc
Prevent depacketizer OOB reads on zero-length VP8 payload.
2015-06-30 09:12:09 +00:00
rtp_format_vp8.cc
Use webrtc/base/logging.h for rtp_rtcp.
2015-10-28 15:39:43 +00:00
rtp_format_vp8.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format_vp9_unittest.cc
Reland of "Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile."
2015-11-06 08:12:09 +00:00
rtp_format_vp9.cc
Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
2015-11-10 15:17:26 +00:00
rtp_format_vp9.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_format.cc
Add support for VP9 packetization/depacketization.
2015-07-28 11:02:58 +00:00
rtp_format.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_header_extension_unittest.cc
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_header_extension.cc
Enable CVO by default through webrtc pipeline.
2015-04-01 22:33:15 +00:00
rtp_header_extension.h
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
rtp_header_parser.cc
[rtp_rtcp] Lint errors cleaned from rtp_utility
2015-12-28 18:18:52 +00:00
rtp_packet_history_unittest.cc
Removed vie_defines.h
2015-11-06 12:33:56 +00:00
rtp_packet_history.cc
lint build/include errors fixed in rtp_rtcp module
2015-12-10 13:05:35 +00:00
rtp_packet_history.h
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
rtp_payload_registry_unittest.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
rtp_payload_registry.cc
Remove video-codec max bitrate from TMMBN.
2016-02-16 16:59:36 +00:00
rtp_receiver_audio.cc
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_audio.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_impl.cc
Remove deprecated RtpReceiver::CreateAudioReceiver() function.
2016-04-01 08:01:33 +00:00
rtp_receiver_impl.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_strategy.cc
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
rtp_receiver_strategy.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_receiver_video.cc
Deprecate VideoDecoder::Reset() and remove calls.
2016-02-02 14:40:13 +00:00
rtp_receiver_video.h
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
rtp_rtcp_config.h
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00
rtp_rtcp_impl_unittest.cc
Remove webrtc::ScopedVector
2016-03-31 17:24:31 +00:00
rtp_rtcp_impl.cc
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtp_rtcp_impl.h
Experiment for the nack module.
2016-03-12 11:30:31 +00:00
rtp_sender_audio.cc
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
2016-03-15 18:41:58 +00:00
rtp_sender_audio.h
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
2016-03-15 18:41:58 +00:00
rtp_sender_unittest.cc
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
2016-03-15 18:41:58 +00:00
rtp_sender_video.cc
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
rtp_sender_video.h
webrtc::RtpPacket name freed for better RtpPacket
2016-02-19 15:02:24 +00:00
rtp_sender.cc
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtp_sender.h
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
rtp_utility.cc
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
rtp_utility.h
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +00:00
ssrc_database.cc
Refactor RtpSender and SSRCDatabase.
2016-02-02 16:34:16 +00:00
ssrc_database.h
Refactor RtpSender and SSRCDatabase.
2016-02-02 16:34:16 +00:00
time_util_unittest.cc
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
time_util.cc
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
time_util.h
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
tmmbr_help.cc
TMMBRSet become vector<rtcp::TmmbItem>
2016-02-29 13:26:05 +00:00
tmmbr_help.h
TMMBRSet become vector<rtcp::TmmbItem>
2016-02-29 13:26:05 +00:00
video_codec_information.h
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00
vp8_partition_aggregator_unittest.cc
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
vp8_partition_aggregator.cc
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00
vp8_partition_aggregator.h
in rtp_rtcp module:
2015-12-10 20:39:16 +00:00