zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
9e69dfdfd5f38cdbd335280fb158a6e2a11a65d6
platform-external-webrtc/webrtc
History
magjed 9e69dfdfd5 Java SurfaceTextureHelper: Remove support for external thread 
Currently, VideoCapturerAndroid owns a dedicated tread, and
SurfaceTextureHelper get this thread passed in the ctor. In
VideoCapturerAndroid.dispose(), ownership of the thread is passed to
SurfaceTextureHelper so that we can return directly instead of waiting
for the last frame to return.

This CL makes the SurfaceTextureHelper own the thread the whole time
instead, and VideoCapturerAndroid calls getHandler() to get it instead.

BUG=webrtc:5519

Review URL: https://codereview.webrtc.org/1738123002

Cr-Commit-Position: refs/heads/master@{#11790}
2016-02-26 15:45:50 +00:00
..
api
Java SurfaceTextureHelper: Remove support for external thread
2016-02-26 15:45:50 +00:00
audio
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +00:00
base
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +00:00
build
MB: Fix typo in device mixin.
2016-02-25 07:41:17 +00:00
call
Break out pacer thread from CongestionController to increase testability.
2016-02-24 15:03:08 +00:00
common_audio
Replace scoped_ptr with unique_ptr in webrtc/common_audio/
2016-02-24 13:22:40 +00:00
common_video
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
2016-02-26 15:38:57 +00:00
examples
Make VideoTrack and VideoTrackRenderers implement rtc::VideoSourceInterface.
2016-02-26 09:25:02 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
2016-02-26 15:38:57 +00:00
modules
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
2016-02-26 15:38:57 +00:00
p2p
Fix bug 574524: DtlsTransportChannel crashes after SSL closes remotely
2016-02-24 19:10:09 +00:00
pc
Replace scoped_ptr with unique_ptr in webrtc/media/
2016-02-26 11:00:39 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Integrate helper macros for calling histograms with different names (real-time vs screenshare and rampup metrics).
2016-02-04 08:33:29 +00:00
test
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
2016-02-26 15:38:57 +00:00
tools
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
2016-02-19 15:04:56 +00:00
video
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
2016-02-26 15:38:57 +00:00
voice_engine
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +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
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +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
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +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
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Move RTP module activation into PayloadRouter.
2016-02-26 15:31:44 +00:00
common.gyp
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +00:00
common.h
Added boilerplate code for being able to test the upcoming
2016-02-17 09:11:24 +00:00
config.cc
Support multiple rtx codecs.
2016-02-03 12:30:10 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +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
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
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
Remove no-op VideoDecoder::Reset implementation.
2016-02-12 14:16:31 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +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
Revert of Cleanup of webrtc::VideoFrame. (patchset #6 id:100001 of https://codereview.webrtc.org/1679323002/ )
2016-02-26 15:38:57 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.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
video_send_stream.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
webrtc_examples.gyp
Move RTCFileLogger to webrtc/base/objc.
2016-02-25 00:33:22 +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
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +00:00
webrtc.gyp
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
2016-02-25 16:37:02 +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.