platform-external-webrtc

Files

peah c54aad6ae0 Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )

Reason for revert:
This CL caused a google3 breakage due to dependencies in Google3.

I will fix that, and reland.

Original issue's description:
> Moved ring-buffer related files from common_audio to audio_processing
>
> BUG=webrtc:5724
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/711ccc8d96490f58cc3d7fd9207c19d4d881d4dc
> Cr-Commit-Position: refs/heads/master@{#12227}

TBR=ivoc@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5724

Review URL: https://codereview.webrtc.org/1856323002

Cr-Commit-Position: refs/heads/master@{#12232}

2016-04-05 07:02:35 +00:00

api

Add CoreVideoFrameBuffer.

2016-04-04 21:10:47 +00:00

audio

Reland "Add check_deps rules in DEPS files."

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

base

Improve iOS frame capture threading.

2016-04-01 00:14:09 +00:00

build

Revert of Set defines for Chromium build. (patchset #3 id:40001 of https://codereview.webrtc.org/1847013002/ )

2016-04-05 06:39:51 +00:00

call

Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).

2016-03-29 09:15:11 +00:00

common_audio

Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )

2016-04-05 07:02:35 +00:00

common_video

Add CoreVideoFrameBuffer.

2016-04-04 21:10:47 +00:00

examples

Deprecate GetWidth() and GetHeight() methods. Replaced by width() and height().

2016-04-04 07:57:37 +00:00

libjingle

Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )

2016-03-31 14:23:52 +00:00

media

Revert of Set defines for Chromium build. (patchset #3 id:40001 of https://codereview.webrtc.org/1847013002/ )

2016-04-05 06:39:51 +00:00

modules

Revert of Moved ring-buffer related files from common_audio to audio_processing (patchset #8 id:150001 of https://codereview.webrtc.org/1846903004/ )

2016-04-05 07:02:35 +00:00

p2p

Add the last_sent_packet_id to the candidate pair change signal

2016-03-31 19:37:40 +00:00

Add the last_sent_packet_id to the candidate pair change signal

2016-03-31 19:37:40 +00:00

system_wrappers

Remove webrtc::ScopedVector

2016-03-31 17:24:31 +00:00

test

Remove webrtc::ScopedVector

2016-03-31 17:24:31 +00:00

tools

Reland "Add check_deps rules in DEPS files."

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

video

Replace NULL with nullptr in webrtc/video.

2016-04-04 15:56:22 +00:00

voice_engine

- Add temporary VoEBase::audio_device_module() method.

2016-03-31 06:28:56 +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

Make the audio channel communicate network state changes to the call.

2016-03-22 22:32:31 +00:00

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +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

Delete class webrtc::VideoRenderer and its header file.

2016-03-23 11:48:17 +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_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

Delete class webrtc::VideoRenderer and its header file.

2016-03-23 11:48:17 +00:00

video_send_stream.h

Delete class webrtc::VideoRenderer and its header file.

2016-03-23 11:48:17 +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

…

webrtc_perf_tests.isolate

…

webrtc_tests.gypi

Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )

2016-03-31 14:23:52 +00:00

webrtc.gyp

GYP: Add webrtc/pc/pc.gyp:* to 'All' target.

2016-04-01 09:15:05 +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.