webrtc

History

Jelena Marusic 06b08afb67 VoE: VoEBase unit test

Changes:
1. Documented return values of VoEBase methods.
2. Added/modified VoEBase unit test.

R=henrika@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51669004

Cr-Commit-Position: refs/heads/master@{#9038}

2015-04-21 09:39:41 +00:00

base

rtc::Buffer: Accept void* in addition to the byte-sized types

2015-04-20 20:55:41 +00:00

build

Move target_subarch from gyp_webrtc to supplement.gypi

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

common_audio

Fix unknown option '-msse2' warning

2015-04-16 06:57:12 +00:00

common_video

Remove i420_video_frame.h from common_video.gyp

2015-04-10 12:40:21 +00:00

examples/android

Remove henrike@ from OWNERS

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

libjingle

Do not define POSIX.

2015-04-20 11:14:22 +00:00

modules

rtc::Buffer improvements

2015-04-20 12:03:00 +00:00

overrides

Cleanup: Remove DISALLOW_EVIL_CONSTRUCTORS macro.

2015-04-04 23:56:56 +00:00

p2p

rtc::Buffer improvements

2015-04-20 12:03:00 +00:00

sound

Remove henrike@ from OWNERS

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

system_wrappers

Remove android_webview_build conditions.

2015-04-09 15:36:13 +00:00

test

Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream).

2015-04-15 16:00:37 +00:00

tools

VoE: move mock directory 1 level up

2015-04-15 10:45:09 +00:00

video

Deliver RTCP packets only once per receive stream.

2015-04-16 13:59:35 +00:00

video_engine

Remove simulcast modules from ViEReceiver.

2015-04-17 13:13:52 +00:00

voice_engine

VoE: VoEBase unit test

2015-04-21 09:39:41 +00:00

.gitignore

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

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

BUILD.gn

Let Chromium declare the mips_dsp_rev build variable.

2015-03-04 09:51:17 +00:00

call.h

Remove ViE interface usage from VideoReceiveStream.

2015-04-09 12:35:46 +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

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Fix style violations in common_types.h and config.h

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

config.h

Add CVO support to Vie layer.

2015-03-12 20:51:50 +00:00

engine_configurations.h

Delete all codec-specific subclasses of ACMGenericCodec

2015-02-23 09:26:51 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

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

frame_callback.h

Changing include guard in frame_callback.h.

2015-02-03 14:51:39 +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

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

video_decoder.h

Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."

2015-03-16 13:48:18 +00:00

video_encoder.h

Remove default arguments in EncodedImageCallback.

2015-02-09 09:14:48 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""

2015-04-09 11:44:19 +00:00

video_receive_stream.h

Add decoder-timing stats to VideoReceiveStream.

2015-02-25 10:42:45 +00:00

video_renderer.h

Use VideoReceiveStream as an ExternalRenderer.

2015-02-09 15:15:24 +00:00

video_send_stream.h

Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""

2015-03-18 09:51:44 +00:00

webrtc_examples.gyp

Remove video from WebRTC Android example.

2015-03-25 13:32:30 +00:00

webrtc_perf_tests.isolate

Offline screenshare quality test, plus loopback.

2015-02-18 12:46:44 +00:00

webrtc_tests.gypi

Add a BWE fairness test.

2015-04-16 18:20:26 +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.