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webrtc_m130/audio
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Alex Narest cedd351e73 Do not add audio bitrate observer if TWCC sending is not supported 
Bug: webrtc:8243
Change-Id: Ida076dca72a6894053bdd0884f818ab3eaf5128a
Reviewed-on: https://webrtc-review.googlesource.com/30840
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21149}
2017-12-07 20:46:38 +00:00
..
test
Use the new AudioProcessing statistics everywhere.
2017-11-24 18:17:39 +00:00
utility
Stop using public_deps in audio/utility.
2017-12-05 13:52:12 +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
Do not add audio bitrate observer if TWCC sending is not supported
2017-12-07 20:46:38 +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
…
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