signalapp/webrtc
signalapp/webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
3ff50fba59
webrtc/audio
History
Mirko Bonadei 75df7282eb Revert "Break up rtc_event_log_api to solve circular dependencies." 
This reverts commit 001546da95.

Reason for revert: breaks downstream projects.

Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
> 
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
> 
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}

TBR=phoglund@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com

Change-Id: I82540eac176c4abfb7e50dc51671585b32a1bace
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/46581
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21823}
2018-01-31 09:39:44 +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_level.cc Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
audio_level.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
audio_receive_stream_unittest.cc Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
audio_receive_stream.cc Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
audio_receive_stream.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +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 remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
audio_send_stream.cc Remove logging in audio/* from release builds. 2018-01-25 13:46:54 +00:00
audio_send_stream.h Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
audio_state_unittest.cc Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
audio_state.cc Delete AudioMonitor and related code. 2018-01-30 09:48:29 +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 Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
audio_transport_impl.h Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
BUILD.gn Revert "Break up rtc_event_log_api to solve circular dependencies." 2018-01-31 09:39:44 +00:00
channel_proxy.cc Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
channel_proxy.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
channel.cc Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
channel.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +00:00
conversion.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
mock_voe_channel_proxy.h Delete AudioMonitor and related code. 2018-01-30 09:48:29 +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
remix_resample_unittest.cc Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
remix_resample.cc Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
remix_resample.h Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +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
transport_feedback_packet_loss_tracker_unittest.cc Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
transport_feedback_packet_loss_tracker.cc Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
transport_feedback_packet_loss_tracker.h Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00