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peah 57d5a2e4df Reland of Added a bitexactness test for the gain controller in the audio processing module. 
This is a reland of the CL https://codereview.webrtc.org/1812433002/ which
was reverted due to incorrect bitexactness on Android bots.

The changes done in the relanding CL is to Deactivate the test for Android and reduce the number of interations.

TBR=henrik.lundin@webrtc.org
BUG=webrtc:5339

Review URL: https://codereview.webrtc.org/1835073004

Cr-Commit-Position: refs/heads/master@{#12143}
2016-03-29 16:48:44 +00:00
..
api
Android HW decoder: Add support for textures when using EGL 1.0
2016-03-29 12:13:29 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Remove accidentally readded webrtc/base/sslstreamadapterhelper.cc
2016-03-29 15:52:27 +00:00
build
Build dynamic framework with podspec for Objective-C API.
2016-03-25 21:49:18 +00:00
call
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
2016-03-29 09:15:11 +00:00
common_audio
Move swap_queue.h to base/
2016-03-24 08:51:59 +00:00
common_video
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
examples
Delete cricket::VideoRenderer.
2016-03-23 11:06:05 +00:00
libjingle
Remove all uses of the HAVE_CONFIG_H define.
2016-03-29 15:47:28 +00:00
media
Remove all uses of the HAVE_CONFIG_H define.
2016-03-29 15:47:28 +00:00
modules
Reland of Added a bitexactness test for the gain controller in the audio processing module.
2016-03-29 16:48:44 +00:00
p2p
Add 16-bit network id to the candidate signaling.
2016-03-23 23:07:54 +00:00
pc
Remove all uses of the HAVE_CONFIG_H define.
2016-03-29 15:47:28 +00:00
system_wrappers
Change include in metrics.h (change to use systems_wrappers/include/logging.h, base logging breaks chromium.fyi).
2016-03-29 11:16:31 +00:00
test
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
2016-03-29 09:15:11 +00:00
voice_engine
Avoid clicks when muting/unmuting a voe::Channel.
2016-03-24 17:36:06 +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
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +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
DEPS
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +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
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +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/sound/ subdir.
2016-03-17 18:23:05 +00:00
webrtc.gyp
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +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.