webrtc_m130

History

Updating full stack test to optionally save metadata for each frame and save it
to a file with given filename (controlled from the new full_stack_samples
executable).
Adding a Python script that reads the output generated by full stack test
and plots the graph(s).

Review URL: https://codereview.webrtc.org/1289933003

Cr-Commit-Position: refs/heads/master@{#9874}

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

base

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

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

build

GN: Fix invalid configuration for Android GCC build.

2015-09-07 10:57:57 +00:00

common_audio

Convert some more things to size_t.

2015-08-29 00:31:15 +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

Fix for frame resolution in encoded frame callback.

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

overrides

Attempt at fixing error on the Chrome Windows FYI bots.

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

p2p

Remove AsyncHttpRequest, AutoPortAllocator, ConnectivityChecker, and HttpPortAllocator.

2015-09-04 11:21:14 +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

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

2015-09-07 07:35:03 +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

Full stack graphs

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

video_engine

Update SendTimeHistory to store complete PacketInfo, not just send time

2015-09-04 11:43:23 +00:00

voice_engine

Stylizing AudioConferenceMixer.

2015-08-31 14:04:44 +00:00

.gitignore

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

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

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

GN: Fix invalid configuration for Android GCC build.

2015-09-07 10:57:57 +00:00

call.h

Add send transports to individual webrtc::Call streams.

2015-08-28 11:07:15 +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

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

stream.h

Define Stream base classes

2015-07-16 07:30:20 +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

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

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

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

Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder

2015-09-04 01:24:53 +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.