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nisse a0591b5473 Delete webrtc::VideoFrame methods buffer and stride. 
To make the HasOneRef/IsMutable hack work, also had to change the
video_frame_buffer method to return a const ref to a scoped_ref_ptr,
to not imply an AddRef.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1900673002
Cr-Commit-Position: refs/heads/master@{#12557}
2016-04-29 09:09:33 +00:00
..
api
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
audio
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +00:00
base
RTCCertificate serialization.
2016-04-28 12:14:30 +00:00
build
Script for concatenating licenses.
2016-04-29 01:02:32 +00:00
call
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +00:00
common_audio
Revert of Avoiding overflow in cross correlation in NetEq. (patchset #6 id:180001 of https://codereview.webrtc.org/1908623002/ )
2016-04-28 09:16:54 +00:00
common_video
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
examples
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
libjingle
Remove the rtc_relative_path GYP variable and similar defines
2016-04-27 17:13:28 +00:00
media
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
modules
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
p2p
Replace scoped_ptr with unique_ptr in webrtc/p2p/
2016-04-27 14:22:58 +00:00
pc
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
sdk
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
system_wrappers
Replace scoped_ptr with unique_ptr in webrtc/base/
2016-04-26 10:13:28 +00:00
test
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
tools
Adding default_exe_manifest to some gn binaries
2016-04-28 07:21:14 +00:00
video
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
voice_engine
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Switch voice transport to use Call and Stream instead of VoENetwork.
2016-04-29 07:57:21 +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
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
2016-04-29 07:58:48 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
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
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +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
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +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
Make pbos owner of additional video files.
2016-02-26 12:29:17 +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_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +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
Delete webrtc::VideoFrame methods buffer and stride.
2016-04-29 09:09:33 +00:00
video_receive_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
video_send_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
webrtc_examples.gyp
Build dynamic iOS SDK.
2016-04-27 08:54:27 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
2016-04-29 07:58:48 +00:00
webrtc.gyp
Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ )
2016-04-29 07:58:48 +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.