platform-external-webrtc

Files

henrikg 8dba03d6d7 Temporarily define RTC_DISALLOW_ASSIGN in Chromium constructormagic override.

The override will be removed shortly in https://codereview.webrtc.org/1342543004/ This is to make the FYI bots happy meanwhile.

BUG=chromium:468375
TBR=tommi@webrtc.org
NOTRY=true

Review URL: https://codereview.webrtc.org/1345193002

Cr-Commit-Position: refs/heads/master@{#9951}

2015-09-16 11:45:31 +00:00

base

Do not print C++ line numbers for Java logging.

2015-09-15 18:05:32 +00:00

build

Android: Enable C99 mode instead of C89 (default).

2015-09-15 06:16:11 +00:00

common_audio

Turn webrtc::Vad into a pure virtual interface

2015-09-09 06:04:57 +00:00

common_video

VideoFrameBuffer: Make non-const data access explicit

2015-08-26 14:06:30 +00:00

examples

Handle all RTCICEConnectionState values in ARDVideoCallViewController

2015-09-01 22:35:42 +00:00

libjingle

Remove webrtc/libjingle/{examples,session}.

2015-06-12 11:13:46 +00:00

modules

Move AudioDecoderOpus next to AudioEncoderOpus

2015-09-15 15:28:29 +00:00

overrides

Temporarily define RTC_DISALLOW_ASSIGN in Chromium constructormagic override.

2015-09-16 11:45:31 +00:00

p2p

Add a rate tracker that tracks rate over a given interval split up into buckets that accumulate unit counts for their portion of said interval and use this instead of the standard rate tracker so that the values of retrieved frame rate stats are completely independent of the polling rate.

2015-09-14 17:38:20 +00:00

sound

Remove henrike@ from OWNERS

2015-04-01 15:08:49 +00:00

system_wrappers

Disable base/logging.h stderr logs by default for webrtc/ tests.

2015-09-07 07:35:03 +00:00

test

Wire up PacketTime to ReceiveStreams.

2015-09-08 12:36:23 +00:00

tools

Update a ton of audio code to use size_t more correctly and in general reduce

2015-08-24 21:52:45 +00:00

video

2015-09-14 17:38:20 +00:00

video_engine

Simplify BitrateAllocator::AddBitrateObserver.

2015-09-15 13:08:12 +00:00

voice_engine

Add a name to the ProcessThread constructor.

2015-09-11 16:52:22 +00:00

.gitignore

…

audio_receive_stream.h

Control combined_audio_video_bwe with config bool.

2015-07-24 14:10:25 +00:00

audio_send_stream.h

Add send transports to individual webrtc::Call streams.

2015-08-28 11:07:15 +00:00

BUILD.gn

Android: Enable C99 mode instead of C89 (default).

2015-09-15 06:16:11 +00:00

call.h

Wire up PacketTime to ReceiveStreams.

2015-09-08 12:36:23 +00:00

codereview.settings

Update deeper codereview.settings files to match the root.

2015-06-24 19:40:39 +00:00

common_types.cc

Separating voice activity flag from audio level in RtpHeaderExtension.

2015-08-10 13:08:46 +00:00

common_types.h

Update a ton of audio code to use size_t more correctly and in general reduce

2015-08-24 21:52:45 +00:00

common.gyp

Fix style violations in common_types.h and config.h

2015-02-26 14:01:28 +00:00

common.h

Make Config::default_value leak instead of having an exit-time destructor.

2015-05-23 00:50:33 +00:00

config.cc

Add support for transport wide sequence numbers

2015-08-03 11:38:48 +00:00

config.h

Add support for transport wide sequence numbers

2015-08-03 11:38:48 +00:00

engine_configurations.h

Remove VideoEngine interfaces.

2015-05-12 14:51:08 +00:00

frame_callback.h

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

libjingle_examples.gyp

Fix build when using Xcode 7 which contains .tbd files instead of .dylib

2015-09-05 03:01:08 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Add the Ooura FFT to RealFourier.

2015-03-19 20:07:43 +00:00

OWNERS

…

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

rtc_unittests.isolate

…

stream.h

Wire up PacketTime to ReceiveStreams.

2015-09-08 12:36:23 +00:00

supplement.gypi

Move target_subarch from gyp_webrtc to supplement.gypi

2015-04-16 07:24:23 +00:00

transport.h

…

typedefs.h

Clear ARM NEON flag

2015-05-20 05:20:04 +00:00

video_decoder.h

iOS HW H264 support.

2015-06-29 21:35:08 +00:00

video_encoder.h

Allow for framerate reduction for HW encoder.

2015-07-13 23:26:40 +00:00

video_engine_tests.isolate

…

video_frame.h

Re-land "Convert native handles to buffers before encoding."

2015-06-05 09:08:12 +00:00

video_receive_stream.h

Wire up currently-received video codec to stats.

2015-08-28 14:35:40 +00:00

video_renderer.h

Rename I420VideoFrame to VideoFrame.

2015-05-30 00:21:56 +00:00

video_send_stream.h

Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/

2015-09-08 12:13:25 +00:00

webrtc_examples.gyp

Adding support for OpenSL ES output in native WebRTC

2015-05-18 14:49:04 +00:00

webrtc_perf_tests.isolate

Enable GoogleWifiTrace3Mbps simulations.

2015-05-04 12:26:26 +00:00

webrtc_tests.gypi

Full stack graphs

2015-09-07 13:04:23 +00:00

webrtc.gyp

Includes webrtc/build/protoc.gypi instead of build/protoc.gypi

2015-07-30 10:45:24 +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.