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ossu 4f43fcfd49 Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility. 
Renamed the new variant of EncodeInternal to EncodeImpl, so that
subclasses implementing one of the EncodeInternal don't need to
explicitly contain 'using AudioEncoder::EncodeInternal' to avoid their
implementation hiding the other variant of EncodeInternal. This causes
a warning (treated as an error) when building using GCC.

Review URL: https://codereview.webrtc.org/1764583003

Cr-Commit-Position: refs/heads/master@{#11868}
2016-03-04 08:54:38 +00:00
..
api
Android SurfaceTextureHelper: Add stopListening() function
2016-03-03 10:18:44 +00:00
audio
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
base
Enable iOS AppRTCDemo send side BWE.
2016-03-04 01:54:37 +00:00
build
Set prefer_fixed_point=1 for the MIPS architecture.
2016-03-03 09:28:56 +00:00
call
Make ReconfigureVideoEncoder void.
2016-03-02 16:00:07 +00:00
common_audio
Replace scoped_ptr with unique_ptr in webrtc/common_audio/
2016-02-24 13:22:40 +00:00
common_video
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
examples
Enable iOS AppRTCDemo send side BWE.
2016-03-04 01:54:37 +00:00
libjingle
Move talk/session/media -> webrtc/pc
2016-02-12 05:48:10 +00:00
media
Change webrtc::VideoSourceInterface to inherit rtc::VideoSourceInterface.
2016-03-03 09:51:56 +00:00
modules
Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.
2016-03-04 08:54:38 +00:00
p2p
Keep on sending stun binding requests on zero-cost networks.
2016-03-03 16:27:53 +00:00
pc
Rename constants files in webrtc/{media,p2p}
2016-03-02 13:42:35 +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
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +00:00
tools
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/
2016-02-19 15:04:56 +00:00
video
Move encoder thread to VideoSendStream.
2016-03-03 15:29:09 +00:00
voice_engine
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +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_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
GN: Update audio_sink.h location
2016-02-29 05:49:54 +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
…
common_types.h
Move RTP stats histograms from VieChannel to SendStatisticsProxy.
2016-03-01 08:40:54 +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
Added boilerplate code for being able to test the upcoming
2016-02-17 09:11:24 +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
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
…
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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +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
Make ReconfigureVideoEncoder void.
2016-03-02 16:00:07 +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
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +00:00
webrtc.gyp
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +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.