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webrtc_m130/webrtc/voice_engine
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nisse 479d3d75df Drop return value from RtpRtcp::IncomingRtcpPacket. 
And from its callee RTCPReceiver::IncomingPacket.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/3009403002
Cr-Commit-Position: refs/heads/master@{#19823}
2017-09-13 14:53:37 +00:00
..
include
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
mock
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
test/auto_test
Update thread annotiation macros to use RTC_ prefix
2017-09-09 11:17:22 +00:00
audio_level.cc
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
2017-07-20 16:57:42 +00:00
audio_level.h
Update thread annotiation macros to use RTC_ prefix
2017-09-09 11:17:22 +00:00
BUILD.gn
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
channel_manager.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channel_manager.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
channel_proxy.cc
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
2017-09-08 15:13:19 +00:00
channel_proxy.h
Add new ANA stats to the old GetStats() to count the number of actions taken by each controller.
2017-09-08 15:13:19 +00:00
channel_unittest.cc
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
channel.cc
Drop return value from RtpRtcp::IncomingRtcpPacket.
2017-09-13 14:53:37 +00:00
channel.h
Update thread annotiation macros to use RTC_ prefix
2017-09-09 11:17:22 +00:00
coder.cc
Fix invalid output buffer usage
2017-04-20 22:19:10 +00:00
coder.h
Move FilePlayer and FileRecorder to Voice Engine
2017-01-10 09:12:51 +00:00
DEPS
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +00:00
file_player_unittests.cc
Replace gflags usages with rtc_base/flags in all targets based on test_main
2017-08-29 12:51:57 +00:00
file_player.cc
Reland "Supporting 48kHz PCM file."
2017-09-01 19:49:24 +00:00
file_player.h
Reland "Supporting 48kHz PCM file."
2017-09-01 19:49:24 +00:00
file_recorder.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
file_recorder.h
Move FilePlayer and FileRecorder to Voice Engine
2017-01-10 09:12:51 +00:00
monitor_module.h
Simplify webrtc::voe::MonitorModule and remove the .cc file.
2017-02-28 09:16:48 +00:00
output_mixer.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
output_mixer.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
shared_data.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
shared_data.h
Update thread annotiation macros to use RTC_ prefix
2017-09-09 11:17:22 +00:00
statistics.cc
Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/
2016-01-21 18:37:44 +00:00
statistics.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
transmit_mixer.cc
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
2017-07-20 16:57:42 +00:00
transmit_mixer.h
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
2017-07-20 16:57:42 +00:00
transport_feedback_packet_loss_tracker_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
transport_feedback_packet_loss_tracker.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
transport_feedback_packet_loss_tracker.h
Move optional.h to webrtc/api/
2017-09-05 15:43:13 +00:00
utility_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
utility.cc
Fix a variable naming typo
2017-07-07 12:29:47 +00:00
utility.h
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
voe_base_impl.cc
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
voe_base_impl.h
External APM usage downstream dependency support cleanup
2017-07-07 11:25:11 +00:00
voe_base_unittest.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
voe_codec_impl.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
voe_codec_impl.h
Regression test for issue where Opus DTX status was being forgotten.
2016-07-27 11:53:52 +00:00
voe_codec_unittest.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
voe_file_impl.cc
Drop unneeded include of media_file.h.
2016-12-21 11:05:44 +00:00
voe_file_impl.h
VoE: apply new style guide on VoE interfaces and their implementations
2015-05-04 12:15:41 +00:00
voe_network_impl.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
voe_network_impl.h
VoE: cleanup VoENetwork implementation
2015-05-06 13:04:29 +00:00
voe_network_unittest.cc
Drop return value from RtpRtcp::IncomingRtcpPacket.
2017-09-13 14:53:37 +00:00
voe_rtp_rtcp_impl.cc
Remove the unused and untested functions from VoERTP_RTCP.
2017-02-14 11:44:57 +00:00
voe_rtp_rtcp_impl.h
Remove the unused and untested functions from VoERTP_RTCP.
2017-02-14 11:44:57 +00:00
voice_engine_defines.h
Delete remnants of RTX support in voice_engine.
2017-08-31 14:26:17 +00:00
voice_engine_fixture.cc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
voice_engine_fixture.h
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
voice_engine_impl.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
voice_engine_impl.h
Remove VoEHardware interface.
2017-03-15 15:08:07 +00:00