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webrtc_m130/webrtc
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Peter Boström b9e77097ed Add QVGA to thresholds for initial quality. 
Makes QualityScaler start at QVGA for <250k initial bitrates. Useful in
combination with overriding max bitrates to a max lower than that for
connections where we know that the max bitrate is capped below where VGA
is useful.

BUG=webrtc:5678
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1900483004 .

Cr-Commit-Position: refs/heads/master@{#12416}
2016-04-18 20:46:06 +00:00
..
api
Fix crash when receiving a texture frame with rotation bit set.
2016-04-18 19:40:28 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
build
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
call
Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo.
2016-04-15 13:34:54 +00:00
common_audio
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
common_video
Added new VideoFrameBuffer methods Data[YUV]() etc.
2016-04-18 12:34:45 +00:00
examples
Use the new appr.tc URL
2016-04-15 14:16:19 +00:00
libjingle
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
media
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
modules
Add QVGA to thresholds for initial quality.
2016-04-18 20:46:06 +00:00
p2p
Make ReliableQuicStream::Write use base::StringPiece, not std::string
2016-04-14 18:12:51 +00:00
pc
Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ )
2016-04-13 17:48:32 +00:00
system_wrappers
Add histogram stats for average QP per frame for VP9 (for sent video streams):
2016-04-18 09:58:52 +00:00
test
Added new VideoFrameBuffer methods Data[YUV]() etc.
2016-04-18 12:34:45 +00:00
tools
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video
Change pre_encode_callback to get a const frame.
2016-04-18 12:15:33 +00:00
voice_engine
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
.gitignore
…
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Update audio_sink.h location
2016-02-29 05:49:54 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
…
common_types.h
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Added support in the AEC for refined filter adaptation.
2016-04-15 18:23:36 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
frame_callback.h
Measure encoding time on encode callbacks.
2016-02-05 10:13:41 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Delete method webrtc::VideoFrame::native_handle.
2016-04-15 10:43:45 +00:00
video_receive_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Change pre_encode_callback to get a const frame.
2016-04-18 12:15:33 +00:00
webrtc_examples.gyp
Add test runner scripts for instrumentation tests
2016-04-08 13:54:32 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc.gyp
GYP: Add webrtc/pc/pc.gyp:* to 'All' target.
2016-04-01 09:15:05 +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.