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skvlad dc1c62cd30 Enable setting the maximum bitrate limit in RtpSender. 
This change allows the application to limit the bitrate of the outgoing
audio and video streams at runtime. The API roughly follows the WebRTC
API draft, defining the RTCRtpParameters structure witn exactly one
encoding (simulcast streams are not exposed in the API for now).
(https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters)

BUG=

Review URL: https://codereview.webrtc.org/1788583004

Cr-Commit-Position: refs/heads/master@{#12025}
2016-03-17 02:07:49 +00:00
..
api
Enable setting the maximum bitrate limit in RtpSender.
2016-03-17 02:07:49 +00:00
audio
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
base
Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
2016-03-16 15:55:48 +00:00
build
Remove references to build_with_libjingle and libjingle_java GYP variables.
2016-03-12 05:34:35 +00:00
call
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
common_audio
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
common_video
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
examples
Remove DeviceManager and DeviceInfo.
2016-03-16 16:35:04 +00:00
libjingle
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
media
Enable setting the maximum bitrate limit in RtpSender.
2016-03-17 02:07:49 +00:00
modules
Dont always downsample to 16kHz in the reverse stream in APM
2016-03-17 01:26:42 +00:00
p2p
Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
2016-03-16 15:55:48 +00:00
pc
Enable setting the maximum bitrate limit in RtpSender.
2016-03-17 02:07:49 +00:00
sound
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
system_wrappers
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
test
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
tools
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
video
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
2016-03-16 00:41:04 +00:00
voice_engine
Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
2016-03-16 14:05:21 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
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
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
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
Changed name for the upcoming AEC from NextGenerationAec to AEC3.
2016-03-08 00:59:43 +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
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
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_media_unittests.isolate
Rename libjingle_media_unittest -> rtc_media_unittests
2016-02-12 14:06:11 +00:00
rtc_pc_unittests.isolate
Rename libjingle_p2p_unittest -> rtc_pc_unittests
2016-02-15 08:36:44 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
2016-02-26 11:21:18 +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.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Cleanup of webrtc::VideoFrame.
2016-02-29 12:11:57 +00:00
video_receive_stream.h
Add histogram stats for AV sync stream offset:
2016-03-15 08:00:54 +00:00
video_renderer.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
video_send_stream.h
Make ReconfigureVideoEncoder void.
2016-03-02 16:00:07 +00:00
webrtc_examples.gyp
Update iOS AppRTCDemo to use the updated Objective-C API.
2016-03-14 05:08:35 +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
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +00:00
webrtc.gyp
Remove webrtc/test/webrtc_test_common.gyp
2016-03-02 18:46:25 +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.