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kwiberg 68d3213313 RTPPayloadRegistry: Stop using the rate to keep track of receive codecs 
It's not used for anything.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2516213002
Cr-Commit-Position: refs/heads/master@{#15438}
2016-12-06 11:52:26 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Revert of Disable P2PTestConductor.LocalP2PTestDtlsBundleInIceRestart under msan (patchset #1 id:1 of https://codereview.webrtc.org/2546913003/ )
2016-12-03 02:09:09 +00:00
audio
Move functionality out from AudioFrame and into AudioFrameOperations.
2016-12-05 09:46:20 +00:00
base
Use RTC_DISALLOW_COPY_AND_ASSIGN in webrtc/base/sigslottester.h
2016-12-05 14:14:34 +00:00
build
Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
2016-12-01 09:45:35 +00:00
call
Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )"
2016-12-05 16:23:48 +00:00
common_audio
Split targets mixing .c and .cc sources.
2016-12-06 06:47:52 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
media
This approach passes packetization mode to the encoder as part of
2016-12-06 10:22:54 +00:00
modules
RTPPayloadRegistry: Stop using the rate to keep track of receive codecs
2016-12-06 11:52:26 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Relaxing timeouts for TestMediaMonitor.
2016-12-01 20:49:25 +00:00
sdk
Expose audio_jitter_buffer_fast_accelerate config to objc wrapper
2016-12-05 22:24:41 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
test
This approach passes packetization mode to the encoder as part of
2016-12-06 10:22:54 +00:00
tools
Added tool for reference less video analysis (go/refless-video-analysis)
2016-12-02 13:00:10 +00:00
video
This approach passes packetization mode to the encoder as part of
2016-12-06 10:22:54 +00:00
voice_engine
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +00:00
.gitignore
…
BUILD.gn
Use RTC_DISALLOW_COPY_AND_ASSIGN in webrtc/base/sigslottester.h
2016-12-05 14:14:34 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
common_types.h
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
2016-12-02 16:58:02 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
…
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +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.