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Per d48015364d Add option to capture to texture in AppRTCDemo for Android. 
The purpose is to be able to easier test and find differences between the path when capturing to textures or byte buffers.

This require https://codereview.webrtc.org/1403713002/ to work.

BUG=webrtc:4993
R=magjed@webrtc.org
TBR=glaznew@webrtc.org

Review URL: https://codereview.webrtc.org/1452423003 .

Cr-Commit-Position: refs/heads/master@{#10766}
2015-11-24 10:13:34 +00:00
..
audio
Move some receive stream configuration into webrtc::AudioReceiveStream.
2015-11-20 17:59:40 +00:00
base
GN: Fix iOS error in audio_device and rtc_base
2015-11-24 01:23:47 +00:00
build
iOS: Set enable_protobuf=1 by default.
2015-11-20 08:37:50 +00:00
call
Add UMA for send bwe and pacer bitrate.
2015-11-17 14:25:02 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Move ThreadWrapper to ProcessThread in base.
2015-11-23 22:48:01 +00:00
examples
Add option to capture to texture in AppRTCDemo for Android.
2015-11-24 10:13:34 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
First part of the preparatory work before the actual work for solving the ducking problem starts.
2015-11-24 07:05:49 +00:00
p2p
Test case for CL 1437933002.
2015-11-23 22:12:44 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Move ThreadWrapper to ProcessThread in base.
2015-11-23 22:48:01 +00:00
test
Move ThreadWrapper to ProcessThread in base.
2015-11-23 22:48:01 +00:00
tools
Remove webrtc/test/channel_transport/include
2015-11-11 09:34:35 +00:00
video
Move ThreadWrapper to ProcessThread in base.
2015-11-23 22:48:01 +00:00
video_engine
Move ThreadWrapper to ProcessThread in base.
2015-11-23 22:48:01 +00:00
voice_engine
Move ThreadWrapper to ProcessThread in base.
2015-11-23 22:48:01 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
audio_state.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
BUILD.gn
Implement fuzzing of VP9 depacketization.
2015-11-23 14:12:13 +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
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
engine_configurations.h
Don't link with audio codecs that we don't use
2015-10-01 04:54:29 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
Add DecodedImageCallback::Decoded() function with custom decode time value.
2015-11-10 13:00:45 +00:00
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
video_receive_stream.h
Require negotiation to send transport cc feedback over RTCP.
2015-11-21 02:05:53 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Remove webrtc/examples/android/media_demo.
2015-11-11 12:11:21 +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
Move BitrateAllocator from BitrateController logic to Call.
2015-11-13 05:02:46 +00:00
webrtc.gyp
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +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.