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834a6ea12bcd65e79c39a8a5cb5e1cd9e13b2ca0
platform-external-webrtc/webrtc
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henrik.lundin 834a6ea12b Add muted_output parameter to ACM 
The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.

A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.

BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
2016-05-13 10:45:31 +00:00
..
api
Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
2016-05-12 19:51:45 +00:00
audio
Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
2016-05-11 13:01:19 +00:00
base
This fixes an issue similar to
2016-05-12 16:28:08 +00:00
build
Revert of Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://cod… (patchset #1 id:1 of https://codereview.webrtc.org/1944273002/ )
2016-05-11 06:28:22 +00:00
call
Refactor before implementing per stream suspension.
2016-05-13 08:43:56 +00:00
common_audio
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
common_video
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
examples
Add magjed@ and perkj@ as webrtc/examples/ owners
2016-05-12 10:48:26 +00:00
libjingle
Fix all -Wnon-virtual-dtor warnings.
2016-05-12 06:11:09 +00:00
media
Fix RTX-configuration test with >2 codecs built.
2016-05-12 14:44:46 +00:00
modules
Add muted_output parameter to ACM
2016-05-13 10:45:31 +00:00
p2p
Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
2016-05-12 19:51:45 +00:00
pc
Refactoring some tests in peerconnectioninterface_unittest.cc.
2016-05-12 15:14:54 +00:00
sdk
Add config continualGatheringPolicy to the IOS RTCConfiguration.
2016-05-11 17:10:47 +00:00
system_wrappers
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
test
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +00:00
tools
Replace scoped_ptr with unique_ptr everywhere
2016-05-01 21:53:55 +00:00
video
Refactor before implementing per stream suspension.
2016-05-13 08:43:56 +00:00
voice_engine
Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
2016-05-11 11:05:13 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
2016-05-11 12:07:33 +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
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +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
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +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
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +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
Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
2016-04-29 09:39:33 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +00:00
webrtc.gyp
New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
2016-05-13 07:43:04 +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.