platform-external-webrtc

Files

henrik.lundin@webrtc.org ed91068bf1 Create a NetEq test for when the first incoming payload type is unknown

This test is currently disabled. It triggers an issue where the NetEq
will trigger asserts on subsequent GetAudio calls if the first inserted
packet is rejected, for instance since the payload type is unknown to
NetEq.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7723 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-11-20 11:01:02 +00:00

base

Annotate COMPILE_ASSERT with __attribute__((unused)).

2014-11-17 13:47:38 +00:00

build

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

common_audio

Update isolate files for Android APK tests.

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

common_video

Update isolate files for Android APK tests.

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

examples

Update Android projects to API level 21.

2014-10-31 23:26:10 +00:00

libjingle

move xmpp and p2p to webrtc

2014-10-28 22:20:11 +00:00

modules

Create a NetEq test for when the first incoming payload type is unknown

2014-11-20 11:01:02 +00:00

overrides

webrtc/overrides: add OWNERS-file.

2014-09-17 08:04:28 +00:00

p2p

Add jmi field for packets discarded due to network error

2014-11-17 19:42:14 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Update isolate files for Android APK tests.

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

test

Change default values for CpuOveruseOptions.

2014-11-20 10:19:46 +00:00

tools

Update isolate files for Android APK tests.

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

video

Change default values for CpuOveruseOptions.

2014-11-20 10:19:46 +00:00

video_engine

Change default values for CpuOveruseOptions.

2014-11-20 10:19:46 +00:00

voice_engine

Report total bitrate for all streams in GetStats.

2014-11-14 11:52:04 +00:00

.gitignore

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

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

BUILD.gn

Change the static_library("webrtc") to a source set in the GN build.

2014-11-12 07:56:21 +00:00

call.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +00:00

common_types.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

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

config.cc

Log formatting fix for VideoEncoderConfig.

2014-11-06 09:35:08 +00:00

config.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

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

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

Move src/ -> webrtc/

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

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +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

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

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

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

typedefs.h

Annotate COMPILE_ASSERT with __attribute__((unused)).

2014-11-17 13:47:38 +00:00

video_decoder.h

Add support for VP9 in webrtc::Call and video_loopback.

2014-11-04 19:41:15 +00:00

video_encoder.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""

2014-09-17 09:02:25 +00:00

video_receive_stream.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Wire up bandwidth stats to the new API and webrtcvideoengine2.

2014-11-05 14:05:29 +00:00

webrtc_examples.gyp

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

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

webrtc_tests.gypi

Put send-side bwe probing under finch experiment.

2014-11-10 13:55:16 +00:00

webrtc.gyp

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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.