platform-external-webrtc

Files

nisse eb83a1a10f This is an initial cleanup step, aiming to delete the

webrtc::VideoRenderer class, replacing it by rtc::VideoSinkInterface.

The next step is to convert all places where a renderer is attached to
rtc::VideoSourceInterface, and at that point, the
SmoothsRenderedFrames method can be replaced by a flag
rtc::VideoSinkWants::smoothed_frames.

Delete unused method IsTextureSupported.
Delete unused time argument to RenderFrame.
Let webrtc::VideoRenderer inherit rtc::VideoSinkInterface. Rename RenderFrame --> OnFrame.

TBR=kjellander@webrtc.org
BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1814763002

Cr-Commit-Position: refs/heads/master@{#12070}

2016-03-21 08:28:06 +00:00

api

Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )

2016-03-20 16:38:44 +00:00

audio

Reland "Add check_deps rules in DEPS files."

2016-03-17 11:05:50 +00:00

base

Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

2016-03-20 13:15:48 +00:00

build

Remove unused libudev on Linux.

2016-03-18 13:08:47 +00:00

call

This is an initial cleanup step, aiming to delete the

2016-03-21 08:28:06 +00:00

common_audio

Reland "Add check_deps rules in DEPS files."

2016-03-17 11:05:50 +00:00

common_video

Reland "Add check_deps rules in DEPS files."

2016-03-17 11:05:50 +00:00

examples

Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ )

2016-03-20 16:38:44 +00:00

libjingle

Reland "Add check_deps rules in DEPS files."

2016-03-17 11:05:50 +00:00

media

This is an initial cleanup step, aiming to delete the

2016-03-21 08:28:06 +00:00

modules

Added a bitexactness test for the echo control mobile in the audio processing module

2016-03-21 08:10:25 +00:00

p2p

Stop using some scoped_ptr features that unique_ptr doesn't have

2016-03-18 05:54:49 +00:00

Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

2016-03-20 13:15:48 +00:00

system_wrappers

Reland "Add check_deps rules in DEPS files."

2016-03-17 11:05:50 +00:00

test

This is an initial cleanup step, aiming to delete the

2016-03-21 08:28:06 +00:00

tools

Reland "Add check_deps rules in DEPS files."

2016-03-17 11:05:50 +00:00

video

This is an initial cleanup step, aiming to delete the

2016-03-21 08:28:06 +00:00

voice_engine

Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API

2016-03-18 03:39:57 +00:00

.gitignore

Flesh out webrtc/.gitignore

2015-11-16 19:02:02 +00:00

audio_receive_stream.h

Replace scoped_ptr with unique_ptr in webrtc/audio/

2016-02-23 18:46:39 +00:00

audio_send_stream.h

Relanding https://codereview.webrtc.org/1715883002/ in pieces.

2016-03-11 11:06:48 +00:00

audio_sink.h

Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."

2016-02-26 21:46:22 +00:00

audio_state.h

Add comments about the Audio parts of the public Call API being WIP.

2015-12-03 12:06:31 +00:00

BUILD.gn

GN: Update audio_sink.h location

2016-02-29 05:49:54 +00:00

call.h

Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().

2015-11-06 23:34:58 +00:00

codereview.settings

Improve informative message in codereview.settings.

2015-11-10 10:08:36 +00:00

common_types.cc

…

common_types.h

Tune BWE to be a bit less sensitive to spurious delay events.

2016-03-10 13:13:26 +00:00

common.gyp

Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."

2016-02-26 21:46:22 +00:00

common.h

Changed name for the upcoming AEC from NextGenerationAec to AEC3.

2016-03-08 00:59:43 +00:00

config.cc

Change transport sequence number extension strings to specify what revision is implemented.

2016-03-01 15:27:27 +00:00

config.h

Re-land: "Use an explicit identifier in Config"

2016-01-14 12:32:51 +00:00

DEPS

This is an initial cleanup step, aiming to delete the

2016-03-21 08:28:06 +00:00

engine_configurations.h

Remove the VoEDtmf interface.

2016-03-08 12:11:00 +00:00

frame_callback.h

Measure encoding time on encode callbacks.

2016-02-05 10:13:41 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

…

OWNERS

Make pbos owner of additional video files.

2016-02-26 12:29:17 +00:00

PATENTS

…

PRESUBMIT.py

…

README.chromium

…

rtc_media_unittests.isolate

Rename libjingle_media_unittest -> rtc_media_unittests

2016-02-12 14:06:11 +00:00

rtc_pc_unittests.isolate

Rename libjingle_p2p_unittest -> rtc_pc_unittests

2016-02-15 08:36:44 +00:00

rtc_unittests.isolate

…

stream.h

…

supplement.gypi

Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac

2016-02-26 11:21:18 +00:00

transport.h

Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )

2015-12-09 11:13:40 +00:00

typedefs.h

Update attributes to match gclibc's ansidecl.h

2016-01-08 12:51:47 +00:00

video_decoder.h

Replace scoped_ptr with unique_ptr in webrtc/common_video/

2016-03-02 11:41:40 +00:00

video_encoder.h

Replace scoped_ptr with unique_ptr in webrtc/common_video/

2016-03-02 11:41:40 +00:00

video_engine_tests.isolate

Add audio streams to CallTest and a first A/V call test.

2016-01-07 16:43:31 +00:00

video_frame.h

Cleanup of webrtc::VideoFrame.

2016-02-29 12:11:57 +00:00

video_receive_stream.h

Add histogram stats for AV sync stream offset:

2016-03-15 08:00:54 +00:00

video_renderer.h

This is an initial cleanup step, aiming to delete the

2016-03-21 08:28:06 +00:00

video_send_stream.h

Make ReconfigureVideoEncoder void.

2016-03-02 16:00:07 +00:00

webrtc_examples.gyp

Update iOS AppRTCDemo to use the updated Objective-C API.

2016-03-14 05:08:35 +00:00

webrtc_nonparallel_tests.isolate

Create isolate files for nonparallel tests.

2015-10-29 14:29:15 +00:00

webrtc_perf_tests.isolate

…

webrtc_tests.gypi

Remove webrtc/sound/ subdir.

2016-03-17 18:23:05 +00:00

webrtc.gyp

Remove webrtc/sound/ subdir.

2016-03-17 18:23:05 +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.