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Per Kjellander 88af20356f Use ProbeClusterConfig in BitrateProber from GoogCC 
Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.

To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>

Deprecates field trial  "WebRTC-Bwe-ProbingConfiguration"

Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36911}
2022-05-17 12:29:25 +00:00
..
adaptation
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
test
Adopt absl::string_view in call/
2022-05-17 12:00:45 +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 GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +00:00
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
audio_send_stream.h
Remove typing detection
2022-03-23 10:23:54 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +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
Async audio processing API
2020-10-02 12:33:34 +00:00
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
2022-03-09 13:23:21 +00:00
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
bitrate_estimator_tests.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
BUILD.gn
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call_config.cc
Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
2021-06-01 06:57:31 +00:00
call_config.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
call_factory.cc
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
call_factory.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
call_perf_tests.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
call.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
degraded_call.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
degraded_call.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
DEPS
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
2021-08-30 10:20:55 +00:00
fake_network_pipe_unittest.cc
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
fake_network_pipe.cc
Delete some unneeded references to ProcessThread.
2022-01-03 15:36:02 +00:00
fake_network_pipe.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
flexfec_receive_stream_impl.cc
Reland: Update local_ssrc without needing to recreate video streams.
2022-05-17 10:59:54 +00:00
flexfec_receive_stream_impl.h
Reland: Update local_ssrc without needing to recreate video streams.
2022-05-17 10:59:54 +00:00
flexfec_receive_stream_unittest.cc
Deduplicate set of the rtp header extension uri constants
2021-09-14 13:38:44 +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
Add a transport_cc() getter and remove rtp_config().
2022-05-09 20:21:14 +00:00
OWNERS
Fix WebRTC version updater bot.
2021-11-22 08:44:15 +00:00
packet_receiver.h
Remove DeliverPacketAsync.
2021-05-29 07:37:33 +00:00
rampup_tests.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rampup_tests.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
receive_stream.h
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
rtp_config.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
rtp_config.h
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
rtp_demuxer_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_demuxer.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
rtp_payload_params.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_payload_params.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_stream_receiver_controller_interface.h
…
rtp_stream_receiver_controller.cc
Remove lock from RtpStreamReceiverController.
2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h
Use SequenceChecker from public API
2021-02-10 15:04:55 +00:00
rtp_transport_config.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_transport_controller_send_factory_interface.h
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
rtp_transport_controller_send_factory.h
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
rtp_transport_controller_send_interface.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_transport_controller_send.cc
Use ProbeClusterConfig in BitrateProber from GoogCC
2022-05-17 12:29:25 +00:00
rtp_transport_controller_send.h
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_video_sender_interface.h
Remove top-level const from parameters in function declarations.
2022-01-26 11:05:25 +00:00
rtp_video_sender_unittest.cc
Remove legacy PacedSender.
2022-05-13 20:31:06 +00:00
rtp_video_sender.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
rtp_video_sender.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.h
IWYU: uint32_t is defined in cstdint
2020-05-07 17:04:15 +00:00
simulated_network_unittest.cc
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
simulated_network.cc
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
simulated_network.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
simulated_packet_receiver.h
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
syncable.cc
…
syncable.h
Make AV sync robust to failures to set a desired minimum delay
2020-09-09 15:44:47 +00:00
version.cc
Update WebRTC code version (2022-05-17T04:04:40).
2022-05-17 05:17:46 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Prepare WebRtcVideoReceiveStream for configuration changes.
2021-07-01 11:23:51 +00:00
video_receive_stream.h
Add a transport_cc() getter and remove rtp_config().
2022-05-09 20:21:14 +00:00
video_send_stream.cc
Change the type of RTCVideoSourceStats.framesPerSecond
2021-11-16 11:21:41 +00:00
video_send_stream.h
Remove more top-level const from parameters in function declarations
2022-02-01 09:15:50 +00:00