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webrtc/audio
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Karl Wiberg e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
..
test Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
utility Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
audio_receive_stream.cc Optional: Use nullopt and implicit construction in /audio 2017-11-17 15:56:17 +00:00
audio_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_receive_stream_unittest.cc Move ADM initialization into WebRtcVoiceEngine 2017-11-21 20:48:07 +00:00
audio_send_stream.cc Optional: Use nullopt and implicit construction in /audio 2017-11-17 15:56:17 +00:00
audio_send_stream.h Deprecated BitrateController::CreateRtcpBandwidthObserver. 2017-11-16 13:52:03 +00:00
audio_send_stream_tests.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
audio_send_stream_unittest.cc Move ADM initialization into WebRtcVoiceEngine 2017-11-21 20:48:07 +00:00
audio_state.cc Move ADM initialization into WebRtcVoiceEngine 2017-11-21 20:48:07 +00:00
audio_state.h Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine. 2017-11-21 10:51:02 +00:00
audio_state_unittest.cc Move ADM initialization into WebRtcVoiceEngine 2017-11-21 20:48:07 +00:00
audio_transport_proxy.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_transport_proxy.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Fix deps of audio:audio_tests. 2017-11-14 08:20:47 +00:00
conversion.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
null_audio_poller.cc Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
null_audio_poller.h Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) 2017-11-01 11:04:26 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
scoped_voe_interface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
time_interval_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00