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webrtc/call
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Mirko Bonadei 675513b96a Stop using LOG macros in favor of RTC_ prefixed macros. 
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
..
test Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_receive_stream.h Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
audio_send_stream.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_send_stream.h Media track ID visibility at BWE level 2017-10-13 13:47:07 +00:00
audio_state.h Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
bitrate_allocator.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
bitrate_allocator.h Reland of BWE allocation strategy 2017-10-20 10:16:15 +00:00
bitrate_allocator_unittest.cc Media track ID visibility at BWE level 2017-10-13 13:47:07 +00:00
bitrate_estimator_tests.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Delete deprecated constructor of SendSideCongestionController. 2017-11-06 15:02:36 +00:00
call.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
call.h Reland of BWE allocation strategy 2017-10-20 10:16:15 +00:00
call_perf_tests.cc Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." 2017-09-29 13:48:29 +00:00
call_unittest.cc Voice Engine: Require caller to supply an AudioDecoderFactory 2017-11-02 13:54:57 +00:00
callfactory.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
callfactory.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
callfactoryinterface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fake_rtp_transport_controller_send.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
flexfec_receive_stream.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
flexfec_receive_stream_impl.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
flexfec_receive_stream_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Remove pbos@webrtc.org from all OWNERS. 2017-11-01 08:03:46 +00:00
rampup_tests.cc Reland of BWE allocation strategy 2017-10-20 10:16:15 +00:00
rampup_tests.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcp_demuxer.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtcp_demuxer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcp_demuxer_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtcp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_config.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_config.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_demuxer.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_demuxer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_demuxer_unittest.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rtp_packet_sink_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtp_stream_receiver_controller.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_transport_controller_send.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtp_transport_controller_send.h Delete deprecated constructor of SendSideCongestionController. 2017-11-06 15:02:36 +00:00
rtp_transport_controller_send_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtx_receive_stream.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtx_receive_stream_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ssrc_binding_observer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
syncable.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_config.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_config.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
video_receive_stream.cc Delete member VideoReceiveStream::Config::Rtp::ulpfec. 2017-09-26 09:49:21 +00:00
video_receive_stream.h Delete member VideoReceiveStream::Config::Rtp::ulpfec. 2017-09-26 09:49:21 +00:00
video_send_stream.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
video_send_stream.h Reland "Add fine grained dropped video frames counters on sending side" 2017-10-25 09:32:15 +00:00