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webrtc_m130/webrtc/modules/rtp_rtcp/source
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asapersson f8cdd184d5 Add histogram stats for AV sync stream offset: 
"WebRTC.Video.AVSyncOffsetInMs"

The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.

Updated sync tests in call_perf_tests.cc to use this implementation.

BUG=webrtc:5493

Review URL: https://codereview.webrtc.org/1756193005

Cr-Commit-Position: refs/heads/master@{#11993}
2016-03-15 08:00:54 +00:00
..
mock
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtcp_packet
[rtp_rtcp] Append functionality moved from base RtcpPacket class to CompoundPacket
2016-03-14 15:19:31 +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
modules: more interface -> include renames
2015-11-04 07:32:04 +00:00
h264_bitstream_parser_unittest.cc
H264 bitstream parser.
2015-09-24 13:07:04 +00:00
h264_bitstream_parser.cc
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +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
Use webrtc/base/logging.h for rtp_rtcp.
2015-10-28 15:39:43 +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
rtt calculation handles time go backwards
2016-03-09 14:14:45 +00:00
rtcp_receiver.h
rtt calculation handles time go backwards
2016-03-09 14:14:45 +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
RTT intermediate calculation use ntp time instead of milliseconds.
2016-02-22 17:59:47 +00:00
rtcp_sender.h
RTT intermediate calculation use ntp time instead of milliseconds.
2016-02-22 17:59:47 +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
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
rtp_receiver_audio.h
lint whitespace warning removed from most rtp_rtcp/source/ files
2015-12-10 10:39:45 +00:00
rtp_receiver_impl.cc
Remove two dead 'using' instances.
2016-01-18 23:24:36 +00:00
rtp_receiver_impl.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtp_receiver_strategy.cc
system_wrappers: rename interface -> include
2015-10-28 17:17:50 +00:00
rtp_receiver_strategy.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_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
RTT intermediate calculation use ntp time instead of milliseconds.
2016-02-22 17:59:47 +00:00
rtp_rtcp_impl.cc
Experiment for the nack module.
2016-03-12 11:30:31 +00:00
rtp_rtcp_impl.h
Experiment for the nack module.
2016-03-12 11:30:31 +00:00
rtp_sender_audio.cc
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtp_sender_audio.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
rtp_sender_unittest.cc
Add transport sequence number on the non-pacer path of the rtp sender.
2016-01-27 11:59:05 +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
Move RTP module activation into PayloadRouter.
2016-02-26 15:31:44 +00:00
rtp_sender.h
Move RTP module activation into PayloadRouter.
2016-02-26 15:31:44 +00:00
rtp_utility.cc
Wire-up BWE feedback for audio receive streams.
2016-01-12 12:55:11 +00:00
rtp_utility.h
[rtp_rtcp] Fix CL#1539423003
2015-12-29 09:12:15 +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