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Peter Boström d7b7ae8bda Add encode/decode time tracing to audio_coding. 
Also removes virtual from VideoDecoder::Decode and updated mocks and
tests accordingly to use VideoDecoder::DecodeInternal instead.

BUG=webrtc:5167
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1512483003 .

Cr-Commit-Position: refs/heads/master@{#10935}
2015-12-08 12:41:44 +00:00
..
audio
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
base
Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ )
2015-12-08 10:32:19 +00:00
build
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +00:00
call
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Nuke webrtc/common_video/plane.*.
2015-12-01 13:10:38 +00:00
examples
Implement standalone event tracing in AppRTCDemo.
2015-12-07 22:17:26 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
Add encode/decode time tracing to audio_coding.
2015-12-08 12:41:44 +00:00
p2p
Add logs when stun or turn host lookup is completed.
2015-12-07 20:06:27 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
2015-12-07 22:29:21 +00:00
test
Create fuzzer tests for audio decoders
2015-12-08 10:27:34 +00:00
tools
Use @webrtc.org addresses for OWNERS.
2015-12-07 18:53:25 +00:00
video
Add test which verifies that the RTP header extensions are set correctly for FEC packets.
2015-12-08 08:10:58 +00:00
voice_engine
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +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
Create fuzzer tests for audio decoders
2015-12-08 10:27:34 +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
Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
2015-11-27 09:02:35 +00:00
common.gyp
…
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
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +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
…
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
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +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
…
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
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +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
Fix Mac ObjC PeerConnection API compilation.
2015-12-03 00:42:41 +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
Merge webrtc/video_engine/ into webrtc/video/
2015-12-08 08:09:07 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.