Danil Chapovalov d673b0fa5d [rtp_rtcp] Fix potentional time difference between rtp and rtcp packets. ...

SetRtpState function was updating only rtp_sender start timestamp.
Now it updates both rtp_sender and rtcp_sender start timestamps.

BUG=webrtc:5433
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1628323003 .

Cr-Commit-Position: refs/heads/master@{#11393}

2016-01-27 11:55:06 +00:00

..

audio_coding

AudioCodingModuleImpl: Put CodecManager and Rent-A-Codec in a separate struct

2016-01-26 14:06:20 +00:00

audio_conference_mixer

Revert of Removing webrtc::AudioFrame::energy_. (patchset #2 id:20001 of https://codereview.webrtc.org/1589953002/ )

2016-01-26 10:29:54 +00:00

audio_device

Roll chromium_revision 7a4fb8d..f527e86 (370025:370073)

2016-01-19 12:47:24 +00:00

audio_processing

Remove mutable from rtc::CriticalSection members.

2016-01-25 11:52:53 +00:00

bitrate_controller

Switch to use new implementation in metrics.h.

2016-01-25 13:58:27 +00:00

desktop_capture

Enable full screen windows to be shown in window picker for mac. Before this patch a full screen window can be shared if sharing is started before the window is entered into full screen mode, but not if it's already in full screen.

2016-01-20 01:34:20 +00:00

include

Revert of Removing webrtc::AudioFrame::energy_. (patchset #2 id:20001 of https://codereview.webrtc.org/1589953002/ )

2016-01-26 10:29:54 +00:00

media_file

Convert channel counts to size_t.

2016-01-13 00:26:55 +00:00

pacing

Temporary hack to avoid assert errors when time moves backwards.

2016-01-22 21:36:39 +00:00

remote_bitrate_estimator

Support REMB in combination with send-side BWE.

2016-01-20 15:14:03 +00:00

rtp_rtcp

[rtp_rtcp] Fix potentional time difference between rtp and rtcp packets.

2016-01-27 11:55:06 +00:00

utility

Remove DISABLED_ON_ macros.

2016-01-04 21:44:16 +00:00

video_capture

Replace hardcoded constant in video capture with macro.

2016-01-19 13:49:25 +00:00

video_coding

H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.

2016-01-27 09:36:07 +00:00

video_processing

Remove mutable from rtc::CriticalSection members.

2016-01-25 11:52:53 +00:00

video_render

Roll chromium_revision 7a4fb8d..f527e86 (370025:370073)

2016-01-19 12:47:24 +00:00

audio_codec_speed_tests.isolate

Update isolate.gypi to support Swarming + move .isolate files

2015-09-25 20:19:21 +00:00

audio_decoder_unittests.isolate

Update isolate.gypi to support Swarming + move .isolate files

2015-09-25 20:19:21 +00:00

audio_device_tests.isolate

Update isolate.gypi to support Swarming + move .isolate files

2015-09-25 20:19:21 +00:00

module_common_types_unittest.cc

modules: more interface -> include renames

2015-11-04 07:32:04 +00:00

modules_java_chromium.gyp

Remove the video capture module on Android.

2015-09-21 08:46:37 +00:00

modules_java.gyp

Remove the video capture module on Android.

2015-09-21 08:46:37 +00:00

modules_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

modules_unittests.isolate

Add new NetEq resources to modules_unittests.isolate.

2016-01-21 12:42:25 +00:00

modules.gyp

H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.

2016-01-27 09:36:07 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00

video_render_tests.isolate

Update isolate.gypi to support Swarming + move .isolate files

2015-09-25 20:19:21 +00:00