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Magnus Jedvert 5e7834e151 Android: Make VideoCapturer an interface for all VideoCapturers to implement 
This CL factors out the interface that AndroidVideoCapturerJni is using to communicate with the Java counterpart. This interface is moved into VideoCapturer. The interface is not touched in this CL, and a follow-up CL is planned to simplify and improve it.

Another change is that the native part of VideoCapturer is created in PeerConnectionFactory.createVideoSource() instead of doing it immediately in the ctor.

BUG=webrtc:5519
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1696553003 .

Cr-Commit-Position: refs/heads/master@{#11606}
2016-02-12 16:05:36 +00:00
..
api
Android: Make VideoCapturer an interface for all VideoCapturers to implement
2016-02-12 16:05:36 +00:00
audio
Clean up of CongestionController.
2016-02-08 13:31:53 +00:00
base
rtc::Buffer: Replace an internal rtc::scoped_ptr with std::unique_ptr
2016-02-11 21:36:57 +00:00
build
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
call
Initialize VideoSendStream members in constructor.
2016-02-11 12:51:18 +00:00
common_audio
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common_video
H264: Improve FFmpeg decoder performance by using I420BufferPool.
2016-02-05 16:08:39 +00:00
examples
Move talk/app/webrtc to webrtc/api
2016-02-10 09:53:26 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Rename libjingle_media_unittest -> rtc_media_unittests
2016-02-12 14:06:11 +00:00
modules
Removes Nexus 5 from AEC and NS blacklists
2016-02-12 15:33:44 +00:00
p2p
Create QuicSession
2016-02-09 01:35:55 +00:00
pc
Add OWNERS file in webrtc/pc
2016-02-12 13:26:58 +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
Added A/V sync tests with drifting clocks.
2016-02-10 18:54:52 +00:00
tools
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
video
Fix race on VCM protection callback.
2016-02-12 15:30:10 +00:00
voice_engine
Remove spammy GetRTPStatistics() log.
2016-02-12 10:35:18 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
2016-02-04 12:12:31 +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
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +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
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
Move gtest_prod_util.h out of webrtc/test tree.
2016-02-05 08:23:57 +00:00
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
libjingle_p2p_unittest.isolate
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
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_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +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
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +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 talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Rename libjingle_media_unittest -> rtc_media_unittests
2016-02-12 14:06:11 +00:00
webrtc.gyp
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +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.