admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/audio
History
Edward Lemur d8b041cc54 Ignore extra arguments in low_bandwidth_audio_test. 
R=phoglund@webrtc.org
TBR=solenberg@webrtc.org

Bug: chromium:755660
Change-Id: I39488f6905875bdc08b006619b972d3f1af2fce1
Reviewed-on: https://webrtc-review.googlesource.com/39924
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21641}
2018-01-16 14:07:54 +00:00
..
test
Ignore extra arguments in low_bandwidth_audio_test.
2018-01-16 14:07:54 +00:00
utility
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
2018-01-10 15:55:04 +00:00
audio_receive_stream_unittest.cc
Delete Channel::OnRecoveredPacket.
2018-01-16 12:21:14 +00:00
audio_receive_stream.cc
Delete Channel::OnRecoveredPacket.
2018-01-16 12:21:14 +00:00
audio_receive_stream.h
Delete Channel::OnRecoveredPacket.
2018-01-16 12:21:14 +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 ownership of voe::Channel into Audio[Receive|Send]Stream.
2018-01-11 12:58:31 +00:00
audio_send_stream.cc
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
2018-01-11 12:58:31 +00:00
audio_send_stream.h
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
2018-01-11 12:58:31 +00:00
audio_state_unittest.cc
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
2018-01-11 12:58:31 +00:00
audio_state.cc
Removes usage of analog AGC.
2018-01-16 13:45:54 +00:00
audio_state.h
Move ownership of voe::Channel into Audio[Receive|Send]Stream.
2018-01-11 12:58:31 +00:00
audio_transport_impl.cc
Removes usage of analog AGC.
2018-01-16 13:45:54 +00:00
audio_transport_impl.h
Removes usage of analog AGC.
2018-01-16 13:45:54 +00:00
BUILD.gn
Make it possible to run low_bandwidth_audio_test on Android swarming.
2018-01-16 11:42:17 +00:00
conversion.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
DEPS
Remove voe::TransmitMixer
2017-12-15 16:48:57 +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
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