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jbauch 944c39006f Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. 
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.

With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.

BUG=webrtc:5155

Review URL: https://codereview.webrtc.org/1785713005

Cr-Commit-Position: refs/heads/master@{#12058}
2016-03-19 08:57:40 +00:00
..
api
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-19 08:57:40 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-19 08:57:40 +00:00
build
Remove unused libudev on Linux.
2016-03-18 13:08:47 +00:00
call
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
common_audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
common_video
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
examples
Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
2016-03-17 22:03:46 +00:00
libjingle
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
media
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-19 08:57:40 +00:00
modules
Enabling rtcp-rsize negotiation and fixing some issues with it.
2016-03-18 22:02:13 +00:00
p2p
Stop using some scoped_ptr features that unique_ptr doesn't have
2016-03-18 05:54:49 +00:00
pc
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
2016-03-19 08:57:40 +00:00
system_wrappers
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
test
Added function for parsing single rtcp packet in tests.
2016-03-17 12:15:20 +00:00
tools
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
video
Enabling rtcp-rsize negotiation and fixing some issues with it.
2016-03-18 22:02:13 +00:00
voice_engine
Deprecate AudioProcessing::AnalyzeReverseStream(AudioFrame) API
2016-03-18 03:39:57 +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
DEPS
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +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
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
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/sound/ subdir.
2016-03-17 18:23:05 +00:00
webrtc.gyp
Remove webrtc/sound/ subdir.
2016-03-17 18:23:05 +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.