platform-external-webrtc

Files

Stefan Holmer 723dff1e6d Poll stats more often to get more stable stats in ramp-up tests.

BUG=chromium:538649
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1384253004 .

Cr-Commit-Position: refs/heads/master@{#10170}

2015-10-05 12:59:50 +00:00

audio

Split webrtc/video into webrtc/{audio,call,video}.

2015-09-25 11:58:39 +00:00

base

Only catch UnsatisfiedLinkError in Logging.java.

2015-10-05 12:27:50 +00:00

build

Don't link with audio codecs that we don't use

2015-10-01 04:54:29 +00:00

call

Add a PacketOptions struct to webrtc::Transport.

2015-10-02 10:39:40 +00:00

common_audio

Add RTC_ prefix to (D)CHECKs and related macros.

2015-09-17 07:24:51 +00:00

common_video

Add RTC_ prefix to (D)CHECKs and related macros.

2015-09-17 07:24:51 +00:00

examples

Loopback and audio only mode.

2015-10-02 18:45:13 +00:00

libjingle

Add RTC_ prefix to contructormagic macros.

2015-09-16 12:37:52 +00:00

modules

Enable HW NS for N6 to fix HW AEC issue

2015-10-05 11:59:01 +00:00

p2p

Reland Change WebRTC SslCipher to be exposed as number only

2015-10-01 04:49:02 +00:00

sound

Add RTC_ prefix to contructormagic macros.

2015-09-16 12:37:52 +00:00

system_wrappers

Add RTC_ prefix to (D)CHECKs and related macros.

2015-09-17 07:24:51 +00:00

test

Add a PacketOptions struct to webrtc::Transport.

2015-10-02 10:39:40 +00:00

tools

Remove AgcManager.

2015-09-29 22:45:23 +00:00

video

Poll stats more often to get more stable stats in ramp-up tests.

2015-10-05 12:59:50 +00:00

video_engine

Move sent key frame stats to send_statistics_proxy class.

2015-10-05 09:36:20 +00:00

voice_engine

Add a PacketOptions struct to webrtc::Transport.

2015-10-02 10:39:40 +00:00

.gitignore

…

audio_receive_stream.h

Align new VoE API with design.

2015-10-01 15:13:46 +00:00

audio_send_stream.h

Align new VoE API with design.

2015-10-01 15:13:46 +00:00

BUILD.gn

Update build files to use webrtc_overrides in Chromium instead of overrides.

2015-09-30 10:49:11 +00:00

call.h

Align new VoE API with design.

2015-10-01 15:13:46 +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

Unify newapi::RtcpMode and RTCPMethod.

2015-10-02 09:37:18 +00:00

common.gyp

Fix style violations in common_types.h and config.h

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

common.h

Add RTC_ prefix to contructormagic macros.

2015-09-16 12:37:52 +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

Don't link with audio codecs that we don't use

2015-10-01 04:54:29 +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

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LICENSE_THIRD_PARTY

Add the Ooura FFT to RealFourier.

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

OWNERS

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PATENTS

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PRESUBMIT.py

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README.chromium

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rtc_unittests.isolate

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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

Add a PacketOptions struct to webrtc::Transport.

2015-10-02 10:39:40 +00:00

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

Add RTC_ prefix to (D)CHECKs and related macros.

2015-09-17 07:24:51 +00:00

video_receive_stream.h

Unify newapi::RtcpMode and RTCPMethod.

2015-10-02 09:37:18 +00:00

video_renderer.h

Rename I420VideoFrame to VideoFrame.

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

video_send_stream.h

Unify Transport and newapi::Transport interfaces.

2015-09-28 16:59:36 +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

Remove AgcManager.

2015-09-29 22:45:23 +00:00

webrtc.gyp

Split webrtc/video into webrtc/{audio,call,video}.

2015-09-25 11:58:39 +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.