zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
5f32aca85ef6c2ded2ec89b4b2a49f62b151223a
platform-external-webrtc/webrtc
History
alessiob 5f32aca85e Reland "C++ porting of the initial python script for conversational speech 
generation."

This CL removes the Python script and adds its C++ porting.
The former was in its early stage and it has permanently been removed.

This is a reland of https://codereview.webrtc.org/2740063004/ which
was reverted. Now the build errors are fixed.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2752793002
Cr-Commit-Position: refs/heads/master@{#17300}
2017-03-18 09:29:13 +00:00
..
api
Add CryptoParams to webrtc::MediaSession.
2017-03-18 02:10:37 +00:00
audio
Move delay_based_bwe_ into CongestionController
2017-03-09 14:40:08 +00:00
base
Revert of Reland: Use native (optimized) functions for byte order conversion. (patchset #5 id:80001 of https://codereview.webrtc.org/2751403003/ )
2017-03-17 09:28:28 +00:00
call
Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ )
2017-03-17 12:55:25 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Add FullStack test for simulcast screenshare mode.
2017-03-09 17:23:30 +00:00
examples
Add video codec setting to AppRTCMobile on iOS.
2017-03-17 16:01:59 +00:00
logging
Enable GN check for webrtc/logging
2017-03-08 10:12:11 +00:00
media
(Re)move VoE specific enums from common_types.h.
2017-03-16 08:20:23 +00:00
modules
Reland "C++ porting of the initial python script for conversational speech
2017-03-18 09:29:13 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
pc
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
sdk
Better handling of error condition in MediaCodecVideoEncoder.
2017-03-15 12:53:14 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete utf_util_win.h.
2017-03-14 08:32:50 +00:00
test
Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ )
2017-03-17 12:55:25 +00:00
tools
Improve error handling for ffmpeg operations
2017-03-15 15:27:31 +00:00
video
Revert of Don't set the priority of the decoder to 'high' on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2745813003/ )
2017-03-17 18:54:25 +00:00
voice_engine
Remove VoEHardware interface.
2017-03-15 15:08:07 +00:00
.gitignore
…
BUILD.gn
Remove voe_cmd_test.
2017-03-07 09:43:21 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
(Re)move VoE specific enums from common_types.h.
2017-03-16 08:20:23 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.