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ivoc caa5f4b3d2 Update to the neteq_rtpplay utility to support RtcEventLog input files. 
This CL adds support for simulating neteq using stored RTP packets as well as calls to GetAudio from an RtcEventLog, using the stored timestamps.
The type of the input file is detected automatically.
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1316903002

Cr-Commit-Position: refs/heads/master@{#9886}
2015-09-08 10:28:53 +00:00
..
base
Remove arraysize.h gcc hack and Chromium override.
2015-09-08 07:57:41 +00:00
build
GN: Fix invalid configuration for Android GCC build.
2015-09-07 10:57:57 +00:00
common_audio
Convert some more things to size_t.
2015-08-29 00:31:15 +00:00
common_video
VideoFrameBuffer: Make non-const data access explicit
2015-08-26 14:06:30 +00:00
examples
Handle all RTCICEConnectionState values in ARDVideoCallViewController
2015-09-01 22:35:42 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Update to the neteq_rtpplay utility to support RtcEventLog input files.
2015-09-08 10:28:53 +00:00
overrides
Remove arraysize.h gcc hack and Chromium override.
2015-09-08 07:57:41 +00:00
p2p
Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer.
2015-09-08 10:12:07 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Disable base/logging.h stderr logs by default for webrtc/ tests.
2015-09-07 07:35:03 +00:00
test
Enabling spatial layers in VP9Impl. Filter layers in the loopback test.
2015-09-08 09:40:36 +00:00
tools
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
video
Enabling spatial layers in VP9Impl. Filter layers in the loopback test.
2015-09-08 09:40:36 +00:00
video_engine
Remove unused overuse detection metric (capture jitter).
2015-09-08 08:52:54 +00:00
voice_engine
Remove unnecessary fields from VoE SharedData.
2015-09-08 09:16:11 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
audio_receive_stream.h
Control combined_audio_video_bwe with config bool.
2015-07-24 14:10:25 +00:00
audio_send_stream.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
BUILD.gn
GN: Fix invalid configuration for Android GCC build.
2015-09-07 10:57:57 +00:00
call.h
Add send transports to individual webrtc::Call streams.
2015-08-28 11:07:15 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +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
Update a ton of audio code to use size_t more correctly and in general reduce
2015-08-24 21:52:45 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
config.h
Add support for transport wide sequence numbers
2015-08-03 11:38:48 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
libjingle_examples.gyp
Fix build when using Xcode 7 which contains .tbd files instead of .dylib
2015-09-05 03:01:08 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Define Stream base classes
2015-07-16 07:30:20 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
Wire up currently-received video codec to stats.
2015-08-28 14:35:40 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder
2015-09-04 01:24:53 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Full stack graphs
2015-09-07 13:04:23 +00:00
webrtc.gyp
Includes webrtc/build/protoc.gypi instead of build/protoc.gypi
2015-07-30 10:45:24 +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.