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0de1c1374c4dba65ade1ec96f839dbd5bc620e3e
platform-external-webrtc/webrtc
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minyue 0de1c1374c Adding DebugDumpReplayer. 
It would be good to have a dedicated DebugDumpReplayer. There is one but it hides itself in DebugDumpTest.

This CL is to separate it out.

BUG=

Review URL: https://codereview.webrtc.org/1810463002

Cr-Commit-Position: refs/heads/master@{#12029}
2016-03-17 09:39:37 +00:00
..
api
Refactor VideoTracks to forward all sinks to its source
2016-03-17 09:35:53 +00:00
audio
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
base
Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
2016-03-16 15:55:48 +00:00
build
Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread.
2016-03-17 09:31:16 +00:00
call
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
common_audio
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
common_video
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
examples
Remove DeviceManager and DeviceInfo.
2016-03-16 16:35:04 +00:00
libjingle
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
media
Refactor VideoTracks to forward all sinks to its source
2016-03-17 09:35:53 +00:00
modules
Adding DebugDumpReplayer.
2016-03-17 09:39:37 +00:00
p2p
Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
2016-03-16 15:55:48 +00:00
pc
Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread.
2016-03-17 09:31:16 +00:00
sound
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
system_wrappers
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
test
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
tools
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
video
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
voice_engine
Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
2016-03-16 14:05:21 +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
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +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
…
common_types.h
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +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
Changed name for the upcoming AEC from NextGenerationAec to AEC3.
2016-03-08 00:59:43 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +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
…
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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +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
Make ReconfigureVideoEncoder void.
2016-03-02 16:00:07 +00:00
webrtc_examples.gyp
Update iOS AppRTCDemo to use the updated Objective-C API.
2016-03-14 05:08:35 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +00:00
webrtc.gyp
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +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.