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platform-external-webrtc/call
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Sebastian Jansson 9129565879 Adds functionality to add delay spikes in SimulatedNetwork. 
Bug: webrtc:9467
Change-Id: Ifddafa65a9e18a3131fc0415764599740fab2db4
Reviewed-on: https://webrtc-review.googlesource.com/92089
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24213}
2018-08-07 16:45:19 +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
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +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
Replaces call config create in tests with modify.
2018-07-13 12:52:45 +00:00
call_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
call.cc
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +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
Adds functionality to add delay spikes in SimulatedNetwork.
2018-08-07 16:45:19 +00:00
fake_network_pipe.h
Adds functionality to add delay spikes in SimulatedNetwork.
2018-08-07 16:45:19 +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
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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Refactoring PayloadRouter.
2018-07-17 14:46:15 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_payload_params_unittest.cc
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_payload_params.cc
Reland "Remove RTPVideoHeader::h264() accessors."
2018-08-03 09:16:50 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_video_sender.cc
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +00:00
rtp_video_sender.h
Rename PayloadRouter to RtpVideoSender.
2018-07-19 08:50:50 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ssrc_binding_observer.h
Delete unneeded includes of basictypes.h.
2018-05-21 19:35:08 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
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