zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
00eb12a20cd29cf249e9ac7cfe0a8037a1e5f903
platform-external-webrtc/audio
History
Niels Möller 61518281a1 Delete always true member voe::Channel::pacing_enabled_ 
Bug: None
Change-Id: If13ea3d2afa6eb149e83cdd179f6bbc7cfabcee9
Reviewed-on: https://webrtc-review.googlesource.com/99500
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24680}
2018-09-11 11:40:11 +00:00
..
test
Mark DirectTransport subclasses ctors as deprecated and switch from them
2018-08-20 12:05:05 +00:00
utility
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
audio_level.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_level.h
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
audio_receive_stream_unittest.cc
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
audio_receive_stream.cc
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
audio_receive_stream.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
audio_send_stream_tests.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_send_stream_unittest.cc
Makes treatment of received reports of packets lost signed.
2018-08-15 14:27:23 +00:00
audio_send_stream.cc
Delete class voe::RtcEventLogProxy
2018-08-15 09:59:15 +00:00
audio_send_stream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_state_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_state.cc
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
audio_state.h
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
audio_transport_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_transport_impl.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
BUILD.gn
Mark DirectTransport subclasses ctors as deprecated and switch from them
2018-08-20 12:05:05 +00:00
channel_proxy.cc
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
channel_proxy.h
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +00:00
channel.cc
Delete always true member voe::Channel::pacing_enabled_
2018-09-11 11:40:11 +00:00
channel.h
Delete always true member voe::Channel::pacing_enabled_
2018-09-11 11:40:11 +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
Refactor voe::Channel to not use RtpReceiver.
2018-08-16 10:18:20 +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
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
OWNERS
…
remix_resample_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remix_resample.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remix_resample.h
Remove dependencies on modules:module_api from AudioProcessing.
2018-04-12 22:05:27 +00:00
time_interval_unittest.cc
Replacing rtc::TimeDelta with webrtc::TimeDelta.
2018-05-08 13:22:53 +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
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
transport_feedback_packet_loss_tracker_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
transport_feedback_packet_loss_tracker.cc
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00
transport_feedback_packet_loss_tracker.h
Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 11:57:00 +00:00