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webrtc/call
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Yves Gerey 6516f76f9b Deprecate SingleThreadedTaskQueueForTesting class. 
This class doesn't strictly follow rtc::TaskQueue semantics,
which makes it surprising and hard to use correctly.
Please use TaskQueueForTest instead.

This CL follows usual deprecation process:

1/ Rename.
% for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i

2/ Annotate old name for downstream users and accidental new uses.

Bug: webrtc:10933
Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29045}
2019-09-03 10:31:30 +00:00
..
test Add RtpPacketPacer interface for pacer control 2019-07-29 15:37:39 +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 Split out RtpSource from libjingle_peerconnection_api 2019-09-02 14:04:47 +00:00
audio_send_stream.cc Introduce MediaTransportConfig 2019-05-21 18:58:33 +00:00
audio_send_stream.h Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo 2019-08-22 07:23:04 +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 [getStats] Implement "media-source" audio levels, fixing Chrome bug. 2019-07-04 08:13:45 +00:00
bitrate_allocator.cc Wire the stable target bitrate from GoogCC to the BitrateAllocator 2019-08-22 15:25:15 +00:00
bitrate_allocator.h Wire the stable target bitrate from GoogCC to the BitrateAllocator 2019-08-22 15:25:15 +00:00
bitrate_allocator_unittest.cc Wire the stable target bitrate from GoogCC to the BitrateAllocator 2019-08-22 15:25:15 +00:00
bitrate_estimator_tests.cc Use explicit TaskQueueFactory for FrameGeneratorCapturer in BitrateEstimatorTest. 2019-04-18 14:17:12 +00:00
BUILD.gn Split out RtpSource from libjingle_peerconnection_api 2019-09-02 14:04:47 +00:00
call.cc Delete audio methods SignalNetworkState 2019-08-30 09:27:30 +00:00
call.h Remove MediaTransport from Call. 2019-08-08 10:58:57 +00:00
call_config.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
call_config.h Make TaskQueueFactory required construction parameter for Call 2019-07-03 14:02:45 +00:00
call_factory.cc DegradedCall: fake network using TaskQueue instead of ProcessThread 2019-08-06 15:05:30 +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 Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
call_unittest.cc Remove MediaTransport from Call. 2019-08-08 10:58:57 +00:00
degraded_call.cc Make fake network degradation work also for sent audio 2019-08-12 15:20:18 +00:00
degraded_call.h Make fake network degradation work also for sent audio 2019-08-12 15:20:18 +00:00
DEPS Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
fake_network_pipe.cc Make fake network degradation work also for sent audio 2019-08-12 15:20:18 +00:00
fake_network_pipe.h Make fake network degradation work also for sent audio 2019-08-12 15:20:18 +00:00
fake_network_pipe_unittest.cc ClangTidy fixes for call/ 2019-03-14 09:38:01 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
flexfec_receive_stream_impl.cc Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc 2019-08-21 09:45:21 +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 Format almost everything. 2019-07-08 13:45:15 +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 Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
rampup_tests.h Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
receive_time_calculator_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtcp_demuxer.cc Format almost everything. 2019-07-08 13:45:15 +00:00
rtcp_demuxer.h Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
rtcp_demuxer_unittest.cc Format almost everything. 2019-07-08 13:45:15 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_config.cc Negotiate use of RTCP loss notification feedback (LNTF) 2019-05-24 12:44:14 +00:00
rtp_config.h Format almost everything. 2019-07-08 13:45:15 +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_demuxer_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
rtp_payload_params.h Reland "Populate the GFD-00 for H264 and generic codecs." 2019-06-14 14:47:06 +00:00
rtp_payload_params_unittest.cc Format almost everything. 2019-07-08 13:45:15 +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 Format almost everything. 2019-07-08 13:45:15 +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 Remove lock for process thread pointer from PacedSender. 2019-08-29 17:08:24 +00:00
rtp_transport_controller_send.h Remove lock for process thread pointer from PacedSender. 2019-08-29 17:08:24 +00:00
rtp_transport_controller_send_interface.h Delete deprecated rtc_event_log header 2019-08-07 10:58:17 +00:00
rtp_video_sender.cc Removes TransportSequenceNumberAllocator 2019-08-28 08:08:37 +00:00
rtp_video_sender.h Delete deprecated rtc_event_log header 2019-08-07 10:58:17 +00:00
rtp_video_sender_interface.h Define FecControllerOverride and plumb it down to VideoEncoder 2019-06-28 15:57:22 +00:00
rtp_video_sender_unittest.cc Disable the most flaky tests on iOS. 2019-08-14 15:42:11 +00:00
rtx_receive_stream.cc Format almost everything. 2019-07-08 13:45:15 +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 Format almost everything. 2019-07-08 13:45:15 +00:00
simulated_network.cc Adds CoDel implementation to network simulation. 2019-02-25 09:54:03 +00:00
simulated_network.h Format almost everything. 2019-07-08 13:45:15 +00:00
simulated_network_unittest.cc Format almost everything. 2019-07-08 13:45:15 +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 Fix frames dropped statistics 2019-08-27 07:43:01 +00:00
video_receive_stream.h Split out RtpSource from libjingle_peerconnection_api 2019-09-02 14:04:47 +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 "Remove the injectable bitrate allocation strategy API." 2019-07-17 10:20:45 +00:00