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webrtc_m130/call
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Sebastian Jansson 7ee2e25afb Extracting the simulation part of FakeNetworkPipe 
This CL extracts the part of FakeNetworkPipe responsible for simulating
network behavior into the SimulatedNetwork class, which implements the
new FakeNetworkInterface.

This prepares for an upcoming CL where the network simulation can
be injected in FakeNetworkPipe, allowing custom simulation models to be
used.

Bug: None
Change-Id: I9b5fa0dd9ff1fd8ccd5a7ce2d9ea3a5b11c5215e
Reviewed-on: https://webrtc-review.googlesource.com/64405
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23146}
2018-05-07 13:59:28 +00:00
..
test
Moving task queue from Call to transport controller.
2018-05-04 12:44:44 +00:00
audio_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_receive_stream.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_send_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
audio_send_stream.h
Wire up MID send value to the PeerConnection API
2018-03-26 18:14:30 +00:00
audio_state.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_state.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
bitrate_allocator_unittest.cc
Uses config struct with bitrate allocator.
2018-04-19 09:15:33 +00:00
bitrate_allocator.cc
Replace runtime thread checks by compile-time checks.
2018-04-27 07:51:55 +00:00
bitrate_allocator.h
Replace runtime thread checks by compile-time checks.
2018-04-27 07:51:55 +00:00
bitrate_constraints.cc
Reland "Moved BitrateConfig out of Call::Config."
2018-02-21 11:38:42 +00:00
bitrate_constraints.h
Reland "Moved BitrateConfig out of Call::Config."
2018-02-21 11:38:42 +00:00
bitrate_estimator_tests.cc
Reland "Move creating encoder to VideoStreamEncoder."
2018-04-19 08:48:58 +00:00
BUILD.gn
Moving task queue from Call to transport controller.
2018-05-04 12:44:44 +00:00
call_config.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
call_config.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
call_perf_tests.cc
Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
2018-04-23 15:31:27 +00:00
call_unittest.cc
Removed fake rtp transport controller send.
2018-02-21 13:39:16 +00:00
call.cc
Moving task queue from Call to transport controller.
2018-05-04 12:44:44 +00:00
call.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
callfactory.cc
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
callfactory.h
Rename Call::Config to CallConfig, keep old name as alias.
2018-02-14 15:14:39 +00:00
degraded_call.cc
Fix race in DegradedCall::DestroyVideoSendStream
2018-03-16 10:23:47 +00:00
degraded_call.h
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Extracting the simulation part of FakeNetworkPipe
2018-05-07 13:59:28 +00:00
fake_network_pipe.h
Extracting the simulation part of FakeNetworkPipe
2018-05-07 13:59:28 +00:00
flexfec_receive_stream_impl.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
flexfec_receive_stream_impl.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream.cc
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
flexfec_receive_stream.h
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
OWNERS
Added nisse@webrtc.org as owner in call.
2018-02-20 09:39:51 +00:00
rampup_tests.cc
Reland "Reland "Move rtp-specific config out of EncoderSettings.""
2018-04-05 14:30:09 +00:00
rampup_tests.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
receive_time_calculator_unittest.cc
Added receive time calculator under field trial.
2018-03-21 15:40:39 +00:00
receive_time_calculator.cc
Fix of ReceiveTimeCalculator field trial parsing.
2018-03-24 12:36:17 +00:00
receive_time_calculator.h
Added receive time calculator under field trial.
2018-03-21 15:40:39 +00:00
rtcp_demuxer_unittest.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtcp_demuxer.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtcp_demuxer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_bitrate_configurator_unittest.cc
Extracted bitrate configuration from call class.
2018-02-21 12:33:02 +00:00
rtp_bitrate_configurator.cc
Extracted bitrate configuration from call class.
2018-02-21 12:33:02 +00:00
rtp_bitrate_configurator.h
Moved bitrate config interface from Call class.
2018-02-21 15:03:45 +00:00
rtp_config.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
rtp_config.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_demuxer_unittest.cc
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
2017-11-22 11:21:47 +00:00
rtp_demuxer.cc
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_demuxer.h
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_packet_sink_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper.cc
Optional: Use nullopt and implicit construction in /call
2018-01-11 12:00:00 +00:00
rtp_rtcp_demuxer_helper.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller_interface.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_stream_receiver_controller.cc
Don't check MIDs when demuxing RTP packets in Call
2018-03-29 20:36:08 +00:00
rtp_stream_receiver_controller.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_transport_controller_send_interface.h
Moving task queue from Call to transport controller.
2018-05-04 12:44:44 +00:00
rtp_transport_controller_send.cc
Moving task queue from Call to transport controller.
2018-05-04 12:44:44 +00:00
rtp_transport_controller_send.h
Moving task queue from Call to transport controller.
2018-05-04 12:44:44 +00:00
rtx_receive_stream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtx_receive_stream.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ssrc_binding_observer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
video_config.cc
Add member VideoEncoderConfig::SdpVideoFormat.
2018-04-17 11:38:23 +00:00
video_config.h
Add member VideoEncoderConfig::SdpVideoFormat.
2018-04-17 11:38:23 +00:00
video_receive_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
video_receive_stream.h
Move videosinkinterface.h to common_video to solve a circular dep.
2018-01-04 13:19:49 +00:00
video_send_stream.cc
Reland "Move creating encoder to VideoStreamEncoder."
2018-04-19 08:48:58 +00:00
video_send_stream.h
Delete unused VideoSendStream::Config::EncoderSettings::encoder.
2018-04-19 13:53:46 +00:00