webrtc

History

Alejandro Luebs f5f8f52a4b Revert "Increase the kMaxNoiseProbability in voice_activity_detector_test"

This reverts commit c9b0f675687d318b9367b1d6764182b9411355de.

It breaks Chromium ASAN.

TBR=niklas.enbom@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1194963003.

Cr-Commit-Position: refs/heads/master@{#9471}

2015-06-19 18:18:02 +00:00

base

Initially when the design was to do this experiment in browser, which doesn't have webrtc code, it requires some glue code to bridge the difference between what's available in webrtc::base and browser process. Now since we're moving to renderer process, we could reuse a lot of existing interfaces instead of rolling our own.

2015-06-18 21:44:46 +00:00

build

Roll chromium_revision e937e5f..c2239a8 (333350:334133)

2015-06-15 09:10:25 +00:00

common_audio

Add missing include guards for audio_ring_buffer.h. Yikes.

2015-06-17 18:39:44 +00:00

common_video

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

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

examples/android/media_demo

VoE: Remove unused interfaces

2015-05-26 08:25:00 +00:00

libjingle

Remove webrtc/libjingle/{examples,session}.

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

modules

Revert "Increase the kMaxNoiseProbability in voice_activity_detector_test"

2015-06-19 18:18:02 +00:00

overrides

Attempt at fixing error on the Chrome Windows FYI bots.

2015-05-25 19:22:23 +00:00

p2p

2015-06-18 21:44:46 +00:00

sound

Remove henrike@ from OWNERS

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

system_wrappers

Reformat existing code. There should be no functional effects.

2015-06-11 21:31:48 +00:00

test

Adds support for webrtc::test::ResourcePath on iOS

2015-06-16 08:04:24 +00:00

tools

Pull the Voice Activity Detector out from the AGC

2015-06-18 19:34:00 +00:00

video

Only use paced packets for estimating bitrate probes.

2015-06-18 14:01:43 +00:00

video_engine

Remove implicit-int-conversion warnings.

2015-06-18 17:00:47 +00:00

voice_engine

Reformat existing code. There should be no functional effects.

2015-06-11 21:31:48 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

audio_receive_stream.h

VoE2 API draft

2015-06-08 11:05:07 +00:00

audio_send_stream.h

VoE2 API draft

2015-06-08 11:05:07 +00:00

BUILD.gn

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

call.h

C++11 in-class member initialization in Call configs.

2015-06-11 10:38:46 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

2014-12-05 13:43:35 +00:00

common_types.cc

Parsing of transport wide sequence number rtp extension header.

2015-03-17 14:33:46 +00:00

common_types.h

Parsing of transport wide sequence number rtp extension header.

2015-03-17 14:33:46 +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 AudioReceiveStream to Call API.

2015-04-29 13:24:10 +00:00

config.h

Add options for NetEq fast accelerate mode through libjingle

2015-06-01 08:29:55 +00:00

engine_configurations.h

Remove VideoEngine interfaces.

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

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Rename I420VideoFrame to VideoFrame.

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

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Add the Ooura FFT to RealFourier.

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

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

supplement.gypi

Move target_subarch from gyp_webrtc to supplement.gypi

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

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Clear ARM NEON flag

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

video_decoder.h

Rename I420VideoFrame to VideoFrame.

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

video_encoder.h

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

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

video_engine_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

video_frame.h

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

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

video_receive_stream.h

C++11 in-class member initialization in Call configs.

2015-06-11 10:38:46 +00:00

video_renderer.h

Rename I420VideoFrame to VideoFrame.

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

video_send_stream.h

C++11 in-class member initialization in Call configs.

2015-06-11 10:38:46 +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

Add HW fallback option to software encoding.

2015-05-19 21:09:17 +00:00

webrtc.gyp

Rename GYP and GN targets for video capture+render.

2015-02-11 07:47:47 +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.