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henrik.lundin 672304a654 NetEq: Remove overly verbose logging 
This change removes all LS_VERBOSE logs that will print once every
packet or more often.

TBR=pbos@webrtc.org
BUG=webrtc:5227

Review URL: https://codereview.webrtc.org/1461903004

Cr-Commit-Position: refs/heads/master@{#10733}
2015-11-20 19:57:11 +00:00
..
audio
Move some receive stream configuration into webrtc::AudioReceiveStream.
2015-11-20 17:59:40 +00:00
base
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
2015-11-20 19:43:27 +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
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
examples
Add OpenSL ES enable setting to AppRTCDemo (part 2).
2015-11-17 18:12:46 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
NetEq: Remove overly verbose logging
2015-11-20 19:57:11 +00:00
p2p
Reland Convert internal representation of Srtp cryptos from string to int
2015-11-19 03:42:00 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
test
modules/video_coding refactorings
2015-11-18 21:00:33 +00:00
tools
Remove webrtc/test/channel_transport/include
2015-11-11 09:34:35 +00:00
video
modules/video_processing: refactor interface->include + more.
2015-11-18 21:31:33 +00:00
video_engine
Allow pacer to boost bitrate in order to meet time constraints.
2015-11-20 17:00:41 +00:00
voice_engine
Remove dead code (we no longer support SILK)
2015-11-19 19:08:35 +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
Remove unused crypto.gni import.
2015-11-04 22:44:04 +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
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +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
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
…
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
…
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
…
video_frame.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
video_receive_stream.h
Fix RTPPayloadRegistry to correctly restore RTX, if a valid mapping exists.
2015-10-14 18:29:56 +00:00
video_renderer.h
…
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
…
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.