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danilchap ac287ee8b5 VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock. 
render_time time field (means capture time for sender side) is used by rtcp SenderReport to calculate offset since last frame and to estimate rtp timestamp for the time SenderReport should be send at.
mapping between rtp timestamp and ntp time in SenderReport is used for stream synchronization.

calculation of rtp_timestamp (using ntp_time of incoming video frame) for rtp packets is unchanged.

BUG=webrtc:5433, webrtc:5504, webrtc:5505

Review URL: https://codereview.webrtc.org/1693443002

Cr-Commit-Position: refs/heads/master@{#11820}
2016-02-29 20:17:10 +00:00
..
api
Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/)
2016-02-29 11:07:45 +00:00
audio
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
base
Added an operator[] to Buffer, to make reading data easier.
2016-02-29 17:36:44 +00:00
build
Make it possible to exclude device management code from rtc_media target.
2016-02-29 14:14:51 +00:00
call
VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock.
2016-02-29 20:17:10 +00:00
common_audio
Replace scoped_ptr with unique_ptr in webrtc/common_audio/
2016-02-24 13:22:40 +00:00
common_video
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
examples
Make VideoTrack and VideoTrackRenderers implement rtc::VideoSourceInterface.
2016-02-26 09:25:02 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Make it possible to exclude device management code from rtc_media target.
2016-02-29 14:14:51 +00:00
modules
Revert "Calculating ERLE in AEC more properly."
2016-02-29 14:20:54 +00:00
p2p
Fix bug 574524: DtlsTransportChannel crashes after SSL closes remotely
2016-02-24 19:10:09 +00:00
pc
Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/)
2016-02-29 11:07:45 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
2016-02-04 08:33:29 +00:00
test
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
tools
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
2016-02-19 15:04:56 +00:00
video
VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock.
2016-02-29 20:17:10 +00:00
voice_engine
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Move RTP module activation into PayloadRouter.
2016-02-26 15:31:44 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Added boilerplate code for being able to test the upcoming
2016-02-17 09:11:24 +00:00
config.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_media_unittests.isolate
Rename libjingle_media_unittest -> rtc_media_unittests
2016-02-12 14:06:11 +00:00
rtc_pc_unittests.isolate
Rename libjingle_p2p_unittest -> rtc_pc_unittests
2016-02-15 08:36:44 +00:00
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Remove no-op VideoDecoder::Reset implementation.
2016-02-12 14:16:31 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
webrtc_examples.gyp
Move RTCFileLogger to webrtc/base/objc.
2016-02-25 00:33:22 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +00:00
webrtc.gyp
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.