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Piotr (Peter) Slatala b275788016 Register stat callbacks after rate observer is registered. 
Currently the stats callback is registered too early.
For now we ignore media transport for these callbacks (it was ignored
already), and we will introduce changes to media transport in the
future.

Bug: webrtc:9719
Bug: chromium:906998
Bug: chromium:906533
Change-Id: I24c0265d46ec2eb35743de6cd96a11d8c41fefbe
Reviewed-on: https://webrtc-review.googlesource.com/c/114904
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26062}
2018-12-19 21:19:01 +00:00
..
test
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +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
Add PeerConnection option to configure minimum audio jitter buffer delay.
2018-11-27 19:49:48 +00:00
audio_send_stream.cc
Remove more unneeded things from ChannelSend
2018-11-21 09:04:07 +00:00
audio_send_stream.h
Remove more unneeded things from ChannelSend
2018-11-21 09:04:07 +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
Don't trigger too many probes when max allocated bitrate changes.
2018-12-11 16:19:53 +00:00
bitrate_allocator.cc
Signals start rate when registering to TargetTransferRateObserver.
2018-12-12 09:22:32 +00:00
bitrate_allocator.h
Signals start rate when registering to TargetTransferRateObserver.
2018-12-12 09:22:32 +00:00
bitrate_estimator_tests.cc
Delete Start and Stop methods from TestVideoCapturer.
2018-12-18 09:29:52 +00:00
BUILD.gn
Merges RtpTransportControllerSend with SendSideCongestionController.
2018-12-06 13:38:39 +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
Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
2018-11-22 11:23:37 +00:00
call_unittest.cc
Remove Transport implementation from ChannelSend
2018-12-18 09:34:52 +00:00
call.cc
Register stat callbacks after rate observer is registered.
2018-12-19 21:19:01 +00:00
call.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
callfactory.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
callfactory.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
degraded_call.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
degraded_call.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
fake_network_pipe.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
flexfec_receive_stream_impl.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
flexfec_receive_stream_impl.h
…
flexfec_receive_stream_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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 unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
rampup_tests.cc
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +00:00
rampup_tests.h
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +00:00
receive_time_calculator_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
rtcp_demuxer_unittest.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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_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
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
rtp_config.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
rtp_config.h
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +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_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
Send and receive color space information if available
2018-12-03 21:07:45 +00:00
rtp_payload_params.cc
Send and receive color space information if available
2018-12-03 21:07:45 +00:00
rtp_payload_params.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +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_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_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
Move RtcpStatistics from common_types.h to a new header file
2018-11-27 13:46:42 +00:00
rtp_transport_controller_send.cc
Cleanup of RtpTransportControllerSend.
2018-12-12 16:36:45 +00:00
rtp_transport_controller_send.h
Cleanup of RtpTransportControllerSend.
2018-12-12 16:36:45 +00:00
rtp_video_sender_interface.h
Move FecController to RtpVideoSender.
2018-10-05 14:39:01 +00:00
rtp_video_sender_unittest.cc
Cleanup of RtpTransportControllerSend.
2018-12-12 16:36:45 +00:00
rtp_video_sender.cc
Remove the audio/video split for the RTCP report intervals.
2018-11-22 01:39:41 +00:00
rtp_video_sender.h
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
2018-11-20 16:28:39 +00:00
rtx_receive_stream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
…
simulated_network.cc
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +00:00
simulated_network.h
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
2018-11-28 18:25:07 +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
…
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
New statistic added to VideoReceiveStream to determine latency to first decode.
2018-12-10 18:49:34 +00:00
video_receive_stream.h
New statistic added to VideoReceiveStream to determine latency to first decode.
2018-12-10 18:49:34 +00:00
video_send_stream.cc
Adding rtcp report interval into RTCConfiguration.
2018-11-12 20:00:00 +00:00
video_send_stream.h
Move RtcpStatistics from common_types.h to a new header file
2018-11-27 13:46:42 +00:00