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peah df80fd1259 When recreating a call based on an aecdump recording the nearend used 
is the one stored in the aecdump.

During AEC development, it is handy to be able to simulate different
doubletalk scenarios. This CL adds the ability to add an
artificial nearend on top of that present in the aecdump, which
allows for the developer to artificially customize the scenario
being tested

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2562593003
Cr-Commit-Position: refs/heads/master@{#15502}
2016-12-09 10:43:45 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
audio
Refactor webrtc/{api,audio} and modules/audio_coding for GN check
2016-12-07 16:23:35 +00:00
base
Clean up redundant include of ../webrtc_overrides
2016-12-07 11:30:50 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
common_audio
Split targets mixing .c and .cc sources.
2016-12-06 06:47:52 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Add FlexFEC settings toggle in Android AppRTCMobile.
2016-12-08 14:51:35 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
2016-12-08 16:04:58 +00:00
modules
When recreating a call based on an aecdump recording the nearend used
2016-12-09 10:43:45 +00:00
p2p
Relanding: Refactoring that removes P2PTransport and DtlsTransport classes.
2016-12-07 00:22:11 +00:00
pc
Relanding: Refactoring that removes P2PTransport and DtlsTransport classes.
2016-12-07 00:22:11 +00:00
sdk
Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
2016-12-08 16:04:58 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add comment to metrics.h
2016-12-06 11:59:08 +00:00
test
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
voice_engine
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +00:00
.gitignore
…
BUILD.gn
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
…
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +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.