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Artem Titov 537b7fed9a Add FakeNetworkPipe ctor from NetworkSimulationInterface. 
Add FakeNetworkPipe ctor from NetworkSimulationInterface to be able to
inject different implementations of simulated networks.

If user requires to do something with injected simulation he can keep
raw reference, or introduce some proxy, that will hide shared ownership
from FakeNetworkPipe. Also FakeNetworkPipe guarantee to keep simulation
object alive while it is alive itself.

Bug: webrtc:9630
Change-Id: Ie00ea1d47bf7658b0c6433cf2efbf364f885c8a0
Reviewed-on: https://webrtc-review.googlesource.com/94153
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24312}
2018-08-16 11:37:38 +00:00
..
test
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +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
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
audio_send_stream.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
audio_send_stream.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bitrate_allocator.h
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
bitrate_estimator_tests.cc
Makes some CallTest members private.
2018-07-13 11:15:45 +00:00
BUILD.gn
Move SimulatedNetwork class to separate file.
2018-08-08 09:29:53 +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
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
call_perf_tests.cc
Revert "Add initial support for RtpEncodingParameters max_framerate."
2018-08-14 07:25:23 +00:00
call_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call.cc
Reland "Refactor RtpVideoStreamReceiver without RtpReceiver."
2018-08-10 08:56:49 +00:00
call.h
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
callfactory.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +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
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
degraded_call.h
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Add FakeNetworkPipe ctor from NetworkSimulationInterface.
2018-08-16 11:37:38 +00:00
fake_network_pipe.h
Add FakeNetworkPipe ctor from NetworkSimulationInterface.
2018-08-16 11:37:38 +00:00
flexfec_receive_stream_impl.cc
replace stringstream in call/
2018-04-04 16:09:15 +00:00
flexfec_receive_stream_impl.h
…
flexfec_receive_stream_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Makes srte owner in call/ and test/
2018-07-13 08:39:41 +00:00
packet_receiver.cc
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
packet_receiver.h
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
rampup_tests.cc
Replaces call config create in tests with modify.
2018-07-13 12:52:45 +00:00
rampup_tests.h
Replaces call config create in tests with modify.
2018-07-13 12:52:45 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
receive_time_calculator.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtcp_demuxer_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtcp_demuxer.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +00:00
rtcp_demuxer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
rtcp_packet_sink_interface.h
…
rtp_bitrate_configurator_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h
Convert video quality test from a TEST_F to a TEST fixture.
2018-06-21 15:49:43 +00:00
rtp_config.cc
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
rtp_config.h
Add shared frame id state to RtpVideoSender.
2018-08-08 15:28:20 +00:00
rtp_demuxer_unittest.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +00:00
rtp_demuxer.cc
Allow all "token" chars from RFC 4566 when checking for legal mid names.
2018-08-01 18:20:42 +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
…
rtp_payload_params_unittest.cc
Remove RTPVideoHeader::vp9() accessors.
2018-08-09 10:53:28 +00:00
rtp_payload_params.cc
Remove RTPVideoHeader::vp9() accessors.
2018-08-09 10:53:28 +00:00
rtp_payload_params.h
Refactoring PayloadRouter.
2018-07-17 14:46:15 +00:00
rtp_rtcp_demuxer_helper_unittest.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.cc
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_rtcp_demuxer_helper.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtp_stream_receiver_controller.h
…
rtp_transport_controller_send_interface.h
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_transport_controller_send.cc
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_transport_controller_send.h
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_video_sender_interface.h
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_video_sender_unittest.cc
Add shared frame id state to RtpVideoSender.
2018-08-08 15:28:20 +00:00
rtp_video_sender.cc
Reland "Refactor RtpVideoStreamReceiver without RtpReceiver."
2018-08-10 08:56:49 +00:00
rtp_video_sender.h
Add shared frame id state to RtpVideoSender.
2018-08-08 15:28:20 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.h
…
simulated_network.cc
Move SimulatedNetwork class to separate file.
2018-08-08 09:29:53 +00:00
simulated_network.h
Use NetworkSimulationInterface instead of SimulatedNetwork.
2018-08-15 10:21:10 +00:00
ssrc_binding_observer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
syncable.cc
…
syncable.h
Replace rtc::Optional with absl::optional in audio, call and video
2018-06-15 12:09:49 +00:00
video_receive_stream.cc
Delete pre_decode_callback.
2018-06-20 07:04:09 +00:00
video_receive_stream.h
Delete pre_decode_callback.
2018-06-20 07:04:09 +00:00
video_send_stream.cc
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00
video_send_stream.h
New api function CreateVideoStreamEncoder.
2018-07-24 09:14:26 +00:00