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kwiberg 686a8efad9 Replace scoped_ptr with unique_ptr in webrtc/media/ 
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1728503002

Cr-Commit-Position: refs/heads/master@{#11779}
2016-02-26 11:00:39 +00:00
..
api
Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ )
2016-02-26 10:54:43 +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
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +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
Replace scoped_ptr with unique_ptr in webrtc/media/
2016-02-26 11:00:39 +00:00
modules
iSAC entropy coder: Avoid signed integer overflow
2016-02-26 10:52:14 +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
…
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
Simple RTCP receiver fuzzer.
2016-02-23 10:35:41 +00:00
tools
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
2016-02-19 15:04:56 +00:00
video
Removes use of DeRegister Rtp Header Extension for video
2016-02-26 10:58:36 +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
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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
…
audio_state.h
…
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
…
codereview.settings
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common_types.cc
…
common_types.h
Move RTCP histograms from vie_channel to video channel stats proxies.
2016-02-24 15:55:06 +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
…
engine_configurations.h
…
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
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OWNERS
…
PATENTS
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PRESUBMIT.py
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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
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stream.h
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supplement.gypi
Roll chromium_revision 1d144ca..fa5d546 (375480:376142)
2016-02-18 19:30:25 +00:00
transport.h
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typedefs.h
…
video_decoder.h
Remove no-op VideoDecoder::Reset implementation.
2016-02-12 14:16:31 +00:00
video_encoder.h
…
video_engine_tests.isolate
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video_frame.h
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video_receive_stream.h
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video_renderer.h
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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
…
webrtc_perf_tests.isolate
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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.