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webrtc/call
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Sebastian Jansson 2b08e3188e Adds CoDel implementation to network simulation. 
Adds an implementation of the CoDel active queue management algorithm
to the network simulation. It is loosely based on CoDel pseudocode
from ACMQueue: https://queue.acm.org/appendices/codel.html

Bug: webrtc:9510
Change-Id: Ice485be35a01dafa6169d697b51b5c1b33a49ba6
Reviewed-on: https://webrtc-review.googlesource.com/c/123581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26834}
2019-02-25 09:54:03 +00:00
..
test Stop using special RTT value for DelayBasedBwe. 2019-02-25 09:51:33 +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 Propagate base minimum delay to audio_receiver_stream 2019-02-06 11:07:42 +00:00
audio_send_stream.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
audio_send_stream.h Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +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 rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +00:00
bitrate_allocator.cc Adds audio priority bitrate field trial parameter. 2019-02-12 16:03:22 +00:00
bitrate_allocator.h Adds audio priority bitrate field trial parameter. 2019-02-12 16:03:22 +00:00
bitrate_allocator_unittest.cc Adds audio priority bitrate field trial parameter. 2019-02-12 16:03:22 +00:00
bitrate_estimator_tests.cc Include absl/memory/memory.h if absl::make_unique is used 2019-01-08 20:08:32 +00:00
BUILD.gn Adds CoDel implementation to network simulation. 2019-02-25 09:54:03 +00:00
call.cc Stop using special RTT value for DelayBasedBwe. 2019-02-25 09:51:33 +00:00
call.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
call_config.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
call_config.h Remove comments about using std::shared_ptr. 2019-01-15 13:15:58 +00:00
call_factory.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
call_factory.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
call_perf_tests.cc Reland "Delete test/constants.h" 2019-02-19 08:51:20 +00:00
call_unittest.cc Remove Transport implementation from ChannelSend 2018-12-18 09:34:52 +00:00
degraded_call.cc Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
degraded_call.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
fake_network_pipe.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
fake_network_pipe_unittest.cc Adds FakeNetworkPipeTest to rtc_unittests. 2019-02-22 15:05:39 +00:00
flexfec_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
flexfec_receive_stream_impl.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +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 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
OWNERS Makes srte owner in call/ and test/ 2018-07-13 08:39:41 +00:00
packet_receiver.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rampup_tests.cc Reland "Always offer transport sequence number header extension for audio"" 2019-02-15 10:57:38 +00:00
rampup_tests.h Refactor bitrate configuration in CallTest 2018-11-13 16:03:00 +00:00
receive_time_calculator.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
receive_time_calculator.h [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
receive_time_calculator_unittest.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
rtcp_demuxer.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtcp_demuxer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_demuxer_unittest.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +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.cc Replace rtc::Optional with absl::optional in audio, call and video 2018-06-15 12:09:49 +00:00
rtp_bitrate_configurator.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +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_config.cc Adding rtcp report interval into RTCConfiguration. 2018-11-12 20:00:00 +00:00
rtp_config.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +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_demuxer_unittest.cc Delete unneeded includes of common_types.h and gn deps on webrtc_common. 2018-11-20 16:28:39 +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.cc H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +00:00
rtp_payload_params.h Pass explicit frame dependency information to RtpPayloadParams 2019-01-29 08:59:48 +00:00
rtp_payload_params_unittest.cc H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +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 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtp_rtcp_demuxer_helper_unittest.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +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 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +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_transport_controller_send.cc Stop using special RTT value for DelayBasedBwe. 2019-02-25 09:51:33 +00:00
rtp_transport_controller_send.h Stop using special RTT value for DelayBasedBwe. 2019-02-25 09:51:33 +00:00
rtp_transport_controller_send_interface.h Stop using special RTT value for DelayBasedBwe. 2019-02-25 09:51:33 +00:00
rtp_video_sender.cc Move FrameCountObserver from RTPSender to RtpVideoSender 2019-01-29 09:31:11 +00:00
rtp_video_sender.h Move FrameCountObserver from RTPSender to RtpVideoSender 2019-01-29 09:31:11 +00:00
rtp_video_sender_interface.h Move FecController to RtpVideoSender. 2018-10-05 14:39:01 +00:00
rtp_video_sender_unittest.cc Update test code to use EncodedImage::Allocate 2019-02-14 15:50:45 +00:00
rtx_receive_stream.cc [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
rtx_receive_stream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtx_receive_stream_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
simulated_network.cc Adds CoDel implementation to network simulation. 2019-02-25 09:54:03 +00:00
simulated_network.h Adds CoDel implementation to network simulation. 2019-02-25 09:54:03 +00:00
simulated_network_unittest.cc Adds CoDel implementation to network simulation. 2019-02-25 09:54:03 +00:00
simulated_packet_receiver.h Calculate next process time in simulated network. 2019-02-08 19:33:17 +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 Enable use of MediaTransportInterface for video streams. 2019-01-11 14:06:15 +00:00
video_receive_stream.h Expose video freeze metrics in GetStats. 2019-02-04 09:58:08 +00:00
video_send_stream.cc (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
video_send_stream.h (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00