platform-external-webrtc

Files

asapersson@webrtc.org cb79141eab Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc.

When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.

Removed unused function ResetRTT.

BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33659005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-18 14:30:32 +00:00

base

Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port

2014-12-18 04:45:05 +00:00

build

Move isolate path into webrtc/build/android/test_runner.py

2014-12-11 11:59:46 +00:00

common_audio

Make safe_conversions suitable for rtc_base_approved.

2014-12-17 22:56:09 +00:00

common_video

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

examples

Update Android projects to API level 21.

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

libjingle

Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."

2014-12-18 02:28:25 +00:00

modules

Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc.

2014-12-18 14:30:32 +00:00

overrides

Remove or rename typedefs with _t prefixes.

2014-12-17 13:43:55 +00:00

p2p

Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port

2014-12-18 04:45:05 +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

Add video send bitrates to histogram stats:

2014-12-09 09:47:53 +00:00

test

Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

2014-12-15 22:09:40 +00:00

tools

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +00:00

video

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

video_engine

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

voice_engine

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

.gitignore

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BUILD.gn

Enabling building with NEON on ARM64

2014-11-26 17:01:40 +00:00

call.h

Wire up RTT statistics to webrtc::Call.

2014-12-11 13:26:09 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

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

common_types.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

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

common.h

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config.cc

Log formatting fix for VideoEncoderConfig.

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

config.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +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

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LICENSE_THIRD_PARTY

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OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

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

PATENTS

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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

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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

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

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

video_frame.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +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

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +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

Add AGC manager tests.

2014-12-16 14:48:47 +00:00

webrtc.gyp

Merge in AGC manager and AGC tools.

2014-12-15 16:33:16 +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.