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sdefresne 1a0c4611d7 Fix compilation of rtc_base_objc on iOS when building as part of Chromium. 
When building as part of Chromium, webrtc/base/logging.h is overridden
by webrtc_overrides/webrtc/base/logging.h that uses Chromium logging
macros.

The implementation of webrtc/base/objc/RTCFileLogger.{h,mm} however has
dependency on the webrtc implementation of logging as it define a class
that inherits from rtc::LogSink. Since the class does not exists when
the overrides are used, the file fails compilation.

Fixes the compilation for iOS as part of Chromium by not compiling the
file webrtc/base/objc/RTCFileLogger.{h,mm} in that configuration.

BUG=459705

Review URL: https://codereview.webrtc.org/1898283002

Cr-Commit-Position: refs/heads/master@{#12438}
2016-04-20 08:45:03 +00:00
..
api
MediaCodecVideoEncoder: Add QP stats to Encoded callback for VP8.
2016-04-20 06:48:26 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Fix compilation of rtc_base_objc on iOS when building as part of Chromium.
2016-04-20 08:45:03 +00:00
build
Fix coverage build.
2016-04-19 06:08:15 +00:00
call
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
common_audio
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
common_video
Delete unused methods SetStartImage and SetTimeoutImage.
2016-04-19 13:19:47 +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
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
modules
Convert Vp8 Rtp headers to frame references.
2016-04-20 08:26:45 +00:00
p2p
Update prflx candidates' generation when setting ICE credentials.
2016-04-20 01:03:35 +00:00
pc
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
system_wrappers
Potential division by zero in RtpToNtpMs() in rtp_to_ntp.cc.
2016-04-19 14:04:52 +00:00
test
Add rotation to EncodedImage and make sure it is passed through encoders.
2016-04-19 13:01:32 +00:00
tools
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video
Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
2016-04-20 08:17:11 +00:00
voice_engine
Add isolate files for Android tests
2016-04-18 03:08:28 +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
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +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
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
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +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
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
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
Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
2016-04-20 08:17:11 +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
Add rotation to EncodedImage and make sure it is passed through encoders.
2016-04-19 13:01:32 +00:00
video_receive_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
video_send_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +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
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc.gyp
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +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.