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danilchap 41befcee7d Make rtcp sender use max transfer unit. 
Remove packet overhead from rtp sender as unused.

R=philipel, åsapersson

Review URL: https://codereview.webrtc.org/1827953002

Cr-Commit-Position: refs/heads/master@{#12165}
2016-03-30 18:11:55 +00:00
..
api
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
build
Build dynamic framework with podspec for Objective-C API.
2016-03-25 21:49:18 +00:00
call
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
2016-03-29 09:15:11 +00:00
common_audio
Move swap_queue.h to base/
2016-03-24 08:51:59 +00:00
common_video
New method I420Buffer::Copy.
2016-03-30 06:44:24 +00:00
examples
Delete cricket::VideoRenderer.
2016-03-23 11:06:05 +00:00
libjingle
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
media
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
modules
Make rtcp sender use max transfer unit.
2016-03-30 18:11:55 +00:00
p2p
Split ByteBuffer into writer/reader objects.
2016-03-30 13:43:44 +00:00
pc
Reset the BWE when the network changes.
2016-03-30 00:27:36 +00:00
system_wrappers
Change include in metrics.h (change to use systems_wrappers/include/logging.h, base logging breaks chromium.fyi).
2016-03-29 11:16:31 +00:00
test
New method I420Buffer::Copy.
2016-03-30 06:44:24 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
2016-03-29 09:15:11 +00:00
voice_engine
Reland https://codereview.webrtc.org/1802993002/
2016-03-30 09:42:37 +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
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +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
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
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_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
…
stream.h
…
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
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +00:00
video_send_stream.h
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +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
…
webrtc_tests.gypi
Remove webrtc/sound/ subdir.
2016-03-17 18:23:05 +00:00
webrtc.gyp
Delete class webrtc::VideoRenderer and its header file.
2016-03-23 11:48:17 +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.