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webrtc_m130/call
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Danil Chapovalov 08fa953711 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" 
This reverts commit fd5166c305068772d00ad7edf50151bba215400b.

Reason for revert: Stop using CreateTestAudioDeviceModule in downstream

Original change's description:
> Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
> 
> This reverts commit fc961357a721cd87dcd45ed409c66cb8cda6f4a2.
> 
> Reason for revert: Breaks downstream importer.
> 
> Original change's description:
> > Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
> > 
> > Bug: webrtc:10284
> > Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28227}
> 
> TBR=danilchap@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: Id6d7571f48771646ddce0f05139a7ea0107759fb
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10284
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141414
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28228}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org,philipel@webrtc.org

Change-Id: I42bc19793d48350ca45b751d7e1b26124ac7fbb9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141670
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28254}
2019-06-12 14:44:01 +00:00
..
test
Removes SendSideCongestionController.
2019-06-11 11:09:24 +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
Introduce MediaTransportConfig
2019-05-21 18:58:33 +00:00
audio_send_stream.cc
Introduce MediaTransportConfig
2019-05-21 18:58:33 +00:00
audio_send_stream.h
[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
2019-05-27 12:40:22 +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_unittest.cc
Removes SendSideCongestionController.
2019-06-11 11:09:24 +00:00
bitrate_allocator.cc
Removes SendSideCongestionController.
2019-06-11 11:09:24 +00:00
bitrate_allocator.h
Enable experiments with audio bitrate priority.
2019-05-08 14:21:01 +00:00
bitrate_estimator_tests.cc
Use explicit TaskQueueFactory for FrameGeneratorCapturer in BitrateEstimatorTest.
2019-04-18 14:17:12 +00:00
BUILD.gn
Removes SendSideCongestionController.
2019-06-11 11:09:24 +00:00
call_config.cc
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
call_config.h
Add base class NetworkPredictor and NetworkPredictorFactory and wire up.
2019-04-10 12:38:58 +00:00
call_factory.cc
ClangTidy fixes for call/
2019-03-14 09:38:01 +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 TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
2019-06-12 14:44:01 +00:00
call_unittest.cc
Introduce MediaTransportConfig
2019-05-21 18:58:33 +00:00
call.cc
Removes SendSideCongestionController.
2019-06-11 11:09:24 +00:00
call.h
Move TaskQueueFactory from Call::Create parameter to CallConfig
2019-04-01 09:27:44 +00:00
degraded_call.cc
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
degraded_call.h
PeerConnection::SetBitrate now also configures media transport.
2019-03-18 19:38:21 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe_unittest.cc
ClangTidy fixes for call/
2019-03-14 09:38:01 +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
flexfec_receive_stream_impl.cc
Modernize RtpRtcp factory function: use unique_ptr as return type
2019-03-06 14:38:39 +00:00
flexfec_receive_stream_impl.h
Injecting Clock in video receive.
2019-03-04 21:53:57 +00:00
flexfec_receive_stream_unittest.cc
Fully qualify googletest symbols.
2019-04-09 17:18:20 +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
OWNERS
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +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
Promote RtcEventLogOutputFile to api/
2019-05-08 12:29:42 +00:00
rampup_tests.h
Refactor bitrate configuration in CallTest
2018-11-13 16:03:00 +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
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
Fully qualify googletest symbols.
2019-04-09 17:18:20 +00:00
rtcp_demuxer.cc
Add functions IsLegalMidName and IsLegalRsidName
2019-03-21 16:10:31 +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
Fully qualify googletest symbols.
2019-04-09 17:18:20 +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_config.cc
Negotiate use of RTCP loss notification feedback (LNTF)
2019-05-24 12:44:14 +00:00
rtp_config.h
Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_sequence_number_map_
2019-05-24 13:02:06 +00:00
rtp_demuxer_unittest.cc
Fully qualify googletest symbols.
2019-04-09 17:18:20 +00:00
rtp_demuxer.cc
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +00:00
rtp_demuxer.h
Log details when RtpDemuxer fails to deliver a packet
2019-04-16 00:47:53 +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
Delete unneeded direct includes of common_types.h
2019-04-01 07:18:13 +00:00
rtp_payload_params.cc
Delete unneeded direct includes of common_types.h
2019-04-01 07:18:13 +00:00
rtp_payload_params.h
Delete unused RTPFragmentationHeader members
2019-05-17 09:26:17 +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
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
(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_transport_controller_send_interface.h
Removes SendSideCongestionController.
2019-06-11 11:09:24 +00:00
rtp_transport_controller_send.cc
Include downlink delay into congestion window size.
2019-05-27 16:07:19 +00:00
rtp_transport_controller_send.h
Include downlink delay into congestion window size.
2019-05-27 16:07:19 +00:00
rtp_video_sender_interface.h
RTPSenderVideo::GetSentRtpPacketInfo() over a set of sequence numbers
2019-04-10 14:32:00 +00:00
rtp_video_sender_unittest.cc
Reland "Cleanup of video packet overhead calculation."
2019-06-10 15:47:48 +00:00
rtp_video_sender.cc
Reland "Cleanup of video packet overhead calculation."
2019-06-10 15:47:48 +00:00
rtp_video_sender.h
Add support for early loss detection using transport feedback.
2019-05-29 13:21:10 +00:00
rtx_receive_stream_unittest.cc
Fully qualify googletest symbols.
2019-04-09 17:18:20 +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
simulated_network_unittest.cc
Adds CoDel implementation to network simulation.
2019-02-25 09:54:03 +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_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
Add jitterBufferDelay and jitterBufferEmittedCount stats for video
2019-05-29 06:23:57 +00:00
video_receive_stream.h
Add jitterBufferDelay and jitterBufferEmittedCount stats for video
2019-05-29 06:23:57 +00:00
video_send_stream.cc
Reland "Inform VideoEncoder of negotiated capabilities"
2019-06-11 14:49:37 +00:00
video_send_stream.h
Reland "Inform VideoEncoder of negotiated capabilities"
2019-06-11 14:49:37 +00:00