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webrtc_m130/call
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Niels Möller cd2e105128 Reenable test RampUpTest.AudioTransportSequenceNumber 
Flakiness should be fixed with cl
https://webrtc-review.googlesource.com/96900

Bug: webrtc:8878
Change-Id: I536d670fdf3b9e52091931e2f37ff9b8d02c2f77
Reviewed-on: https://webrtc-review.googlesource.com/c/110160
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25557}
2018-11-08 12:28:19 +00:00
..
test
Declares BitrateAllocator methods const.
2018-10-24 15:26:36 +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
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
audio_send_stream.cc
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
audio_send_stream.h
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +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
Adds BitrateAllocation struct to OnBitrateUpdated.
2018-10-25 13:51:11 +00:00
bitrate_allocator.cc
Fixes bug in AudioPriorityBitrateAllocationStrategy field trial.
2018-10-25 13:59:22 +00:00
bitrate_allocator.h
Adds BitrateAllocation struct to OnBitrateUpdated.
2018-10-25 13:51:11 +00:00
bitrate_estimator_tests.cc
Add default constructor for rtc::Event
2018-11-07 08:57:50 +00:00
BUILD.gn
Reland "Isolating APM API build target: making :api an actual target."
2018-11-07 14:30:06 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Delete VideoReceiveStream::EnableEncodedFrameRecording
2018-10-02 10:31:46 +00:00
call_perf_tests.cc
Add default constructor for rtc::Event
2018-11-07 08:57:50 +00:00
call_unittest.cc
Use MediaTransportInterface, for audio streams.
2018-10-26 11:40:57 +00:00
call.cc
Move event logging of config into AudioSendStream.
2018-11-02 12:04:24 +00:00
call.h
Removing unnecessary dependencies on socket.h.
2018-10-16 10:24:51 +00:00
callfactory.cc
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
2018-10-08 12:19:31 +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
Adds flags indicating presence in allocation and feedback per packet.
2018-10-09 18:24:38 +00:00
degraded_call.h
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
2018-10-08 12:19:31 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Allow FakeNetworkPipe to wake up its processing thread
2018-10-31 15:20:57 +00:00
fake_network_pipe.h
Allow FakeNetworkPipe to wake up its processing thread
2018-10-31 15:20:57 +00:00
flexfec_receive_stream_impl.cc
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +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
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
flexfec_receive_stream.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
OWNERS
Makes srte owner in call/ and test/
2018-07-13 08:39:41 +00:00
packet_receiver.h
Delete webrtc::PacketTime and backwards compatibility.
2018-08-17 15:14:03 +00:00
rampup_tests.cc
Reenable test RampUpTest.AudioTransportSequenceNumber
2018-11-08 12:28:19 +00:00
rampup_tests.h
Don't send max allocation probe unless allocation changed.
2018-10-16 15:13:57 +00:00
receive_time_calculator_unittest.cc
New method for precise packet reception time measurement.
2018-10-15 11:25:26 +00:00
receive_time_calculator.cc
New method for precise packet reception time measurement.
2018-10-15 11:25:26 +00:00
receive_time_calculator.h
New method for precise packet reception time measurement.
2018-10-15 11:25:26 +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
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
rtp_config.h
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +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
Get frame type, width and height from the generic descriptor.
2018-10-17 13:31:09 +00:00
rtp_payload_params.cc
Get frame type, width and height from the generic descriptor.
2018-10-17 13:31:09 +00:00
rtp_payload_params.h
Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
2018-09-26 10:26:43 +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
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
rtp_transport_controller_send.cc
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
rtp_transport_controller_send.h
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
rtp_video_sender_interface.h
Move FecController to RtpVideoSender.
2018-10-05 14:39:01 +00:00
rtp_video_sender_unittest.cc
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
rtp_video_sender.cc
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
rtp_video_sender.h
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtx_receive_stream.cc
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
2018-09-28 12:00:28 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
simulated_network.cc
Fixing bug in SimulatedNetwork where packets stop.
2018-11-06 19:04:37 +00:00
simulated_network.h
Fixing bug in SimulatedNetwork where packets stop.
2018-11-06 19:04:37 +00:00
simulated_packet_receiver.h
Eliminate methods SetConfig() from DirectTransport and FakeNetworkPipe
2018-08-22 11:12:40 +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
Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
2018-09-28 08:48:02 +00:00
video_receive_stream.h
Enable End-to-End Encrypted Video Frames.
2018-10-18 16:05:13 +00:00
video_send_stream.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
video_send_stream.h
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00