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philipel e0b2f15417 Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer. 
Since we want to stop sending NACKS for frames not needed to keep the stream
decodable we must know which frames that are continuous or not.

BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2322263002 .

Cr-Commit-Position: refs/heads/master@{#14412}
2016-09-28 08:24:02 +00:00
..
androidjunit
Make android_junit_tests use LocalRobolectricTestRunner
2016-09-18 19:28:15 +00:00
api
GN: Fix incorrect include_dir for libjingle_peerconnection_jni target
2016-09-28 07:47:59 +00:00
audio
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
base
Remove unnecessary jsoncpp includes.
2016-09-26 09:52:40 +00:00
build
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
2016-09-28 07:45:12 +00:00
call
Moved Gn target rtc_event_log to one directory above.
2016-09-28 07:49:04 +00:00
common_audio
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
common_video
Add limitations of number of frames that can be created in I420BufferPool::CreateBuffer.
2016-09-27 10:47:48 +00:00
examples
Add usage description strings to Info.plist
2016-09-27 16:02:57 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
media
This is a resubmission of https://codereview.webrtc.org/2047513002/
2016-09-27 10:52:05 +00:00
modules
Frame continuity is now tested as soon as a frame is inserted into the FrameBuffer.
2016-09-28 08:24:02 +00:00
p2p
Delete unused file sessionid.h.
2016-09-27 06:28:51 +00:00
pc
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
sdk
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
stats
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
system_wrappers
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
test
Moved Gn target rtc_event_log to one directory above.
2016-09-28 07:49:04 +00:00
tools
Moved Gn target rtc_event_log to one directory above.
2016-09-28 07:49:04 +00:00
video
Moved Gn target rtc_event_log to one directory above.
2016-09-28 07:49:04 +00:00
voice_engine
Moved Gn target rtc_event_log to one directory above.
2016-09-28 07:49:04 +00:00
.gitignore
…
BUILD.gn
Moved Gn target rtc_event_log to one directory above.
2016-09-28 07:49:04 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Add new decoding statistics for muted output
2016-09-20 08:47:19 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
This is a resubmission of https://codereview.webrtc.org/2047513002/
2016-09-27 10:52:05 +00:00
config.h
This is a resubmission of https://codereview.webrtc.org/2047513002/
2016-09-27 10:52:05 +00:00
DEPS
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Replace VideoCapturerInput with VideoSinkInterface.
2016-09-16 14:53:48 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.