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33fe1dfd3a
webrtc/audio
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Patrik Höglund a8005cfd8b Fix circular dependencies between optional, array_view, and rtc_base. 
This splits things out of rtc_base and makes dependencies explicit.

Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-14 06:49:11 +00:00
..
test Use the new AudioProcessing statistics everywhere. 2017-11-24 18:17:39 +00:00
utility Fix circular dependencies between optional, array_view, and rtc_base. 2017-12-14 06:49:11 +00:00
audio_receive_stream_unittest.cc Move ADM initialization into WebRtcVoiceEngine 2017-11-21 20:48:07 +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_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 Use the new AudioProcessing statistics everywhere. 2017-11-24 18:17:39 +00:00
audio_send_stream.cc Do not add audio bitrate observer if TWCC sending is not supported 2017-12-07 20:46:38 +00:00
audio_send_stream.h Do not add audio bitrate observer if TWCC sending is not supported 2017-12-07 20:46:38 +00:00
audio_state_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_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 circular dependencies between optional, array_view, and rtc_base. 2017-12-14 06:49:11 +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_unittest.cc 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