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platform-external-webrtc/audio
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Sam Zackrisson ff0581672e Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric 
It never saw much use, and is blocking refactoring.

Histograms.xml-side cleanup:
https://chromium-review.googlesource.com/c/chromium/src/+/1344141

Bug: webrtc:7882
Change-Id: I112232a573fcd218dc7a51bfcdd7898847d14f18
Reviewed-on: https://webrtc-review.googlesource.com/c/111506
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25780}
2018-11-26 09:32:35 +00:00
..
test
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
2018-11-26 09:32:35 +00:00
utility
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +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
Delete class ChannelReceiveProxy.
2018-11-16 09:56:54 +00:00
audio_receive_stream.cc
Expose jitter buffer flushes metric in new getStats api.
2018-11-23 11:41:43 +00:00
audio_receive_stream.h
Delete class ChannelSendProxy
2018-11-19 10:17:13 +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
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
2018-11-26 09:32:35 +00:00
audio_send_stream.cc
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
2018-11-26 09:32:35 +00:00
audio_send_stream.h
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
2018-11-26 09:32:35 +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
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
audio_transport_impl.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
BUILD.gn
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
2018-11-26 09:32:35 +00:00
channel_receive.cc
In ReceiveStatistic require callbacks during construction
2018-11-26 09:17:21 +00:00
channel_receive.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
channel_send.cc
Routing BitrateAllocationUpdate to audio codec.
2018-11-22 17:06:52 +00:00
channel_send.h
Routing BitrateAllocationUpdate to audio codec.
2018-11-22 17:06:52 +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
Routing BitrateAllocationUpdate to audio codec.
2018-11-22 17:06:52 +00:00
null_audio_poller.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
null_audio_poller.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
OWNERS
…
remix_resample_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remix_resample.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
remix_resample.h
Remove dependencies on modules:module_api from AudioProcessing.
2018-04-12 22:05:27 +00:00
transport_feedback_packet_loss_tracker_unittest.cc
Cleanup modules_common_types
2018-09-18 08:08:33 +00:00
transport_feedback_packet_loss_tracker.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +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