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04d49500e2469483df67343b403724c2d16d0f45
platform-external-webrtc/call
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Oleh Prypin 04d49500e2 Revert "Using safe casts of allocation limits in Call." 
This reverts commit 4a9b4d6332f596867d2a8fb34ff5b4befb9848eb.

Reason for revert: Breaks downstream projects

Original change's description:
> Using safe casts of allocation limits in Call.
> 
> Bug: None
> Change-Id: I71d0e1f92bf820d117b354dd7701c9c719cc2c0a
> Reviewed-on: https://webrtc-review.googlesource.com/61784
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22490}

TBR=nisse@webrtc.org,srte@webrtc.org

Change-Id: I720e97981574fd152cb7ed4204e29f9ea0b2e909
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/62920
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22491}
2018-03-19 13:29:52 +00:00
..
test
Delete obsolete methods from MockRtpTransportControllerSend
2018-03-19 11:42:11 +00:00
audio_receive_stream.h
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
audio_send_stream.cc
Use rtc::ToString instead of std::to_string
2018-01-29 19:26:09 +00:00
audio_send_stream.h
Move remaining traces of VoiceEngine
2018-01-17 13:27:47 +00:00
audio_state.h
Delete AudioMonitor and related code.
2018-01-30 09:48:29 +00:00
bitrate_allocator_unittest.cc
Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1.
2018-02-28 15:38:15 +00:00
bitrate_allocator.cc
Add a couple of logs.
2018-03-05 16:42:02 +00:00
bitrate_allocator.h
Enables configuration of transmission max bitrate multiplier and fec protection level.
2018-03-02 18:05:29 +00:00
bitrate_constraints.cc
Reland "Moved BitrateConfig out of Call::Config."
2018-02-21 11:38:42 +00:00
bitrate_constraints.h
Reland "Moved BitrateConfig out of Call::Config."
2018-02-21 11:38:42 +00:00
bitrate_estimator_tests.cc
Reland "Add multiplex case to webrtc_perf_tests"
2018-03-10 01:21:04 +00:00
BUILD.gn
Remove the public_deps to fileutils from test_support.
2018-03-16 09:06:27 +00:00
call_perf_tests.cc
Separate test/fake_audio_device on API and implementation. Step 2.
2018-03-12 16:14:39 +00:00
call_unittest.cc
Removed fake rtp transport controller send.
2018-02-21 13:39:16 +00:00
call.cc
Revert "Using safe casts of allocation limits in Call."
2018-03-19 13:29:52 +00:00
call.h
Moved bitrate config interface from Call class.
2018-02-21 15:03:45 +00:00
callfactory.cc
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +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
Fix race in DegradedCall::DestroyVideoSendStream
2018-03-16 10:23:47 +00:00
degraded_call.h
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
DEPS
Make fec controller plug-able.
2018-01-22 11:48:16 +00:00
fake_network_pipe.cc
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
fake_network_pipe.h
Reland: Add ability to emulate degraded network in Call via field trial
2018-03-14 17:03:25 +00:00
flexfec_receive_stream_impl.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +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
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
flexfec_receive_stream.h
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
OWNERS
Added nisse@webrtc.org as owner in call.
2018-02-20 09:39:51 +00:00
rampup_tests.cc
Added some margin to ramp down target in perf test.
2018-02-23 12:15:15 +00:00
rampup_tests.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtcp_demuxer_unittest.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtcp_demuxer.cc
Fix circular dependencies in webrtc_common.
2017-12-15 14:33:26 +00:00
rtcp_demuxer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Extracted bitrate configuration from call class.
2018-02-21 12:33:02 +00:00
rtp_bitrate_configurator.cc
Extracted bitrate configuration from call class.
2018-02-21 12:33:02 +00:00
rtp_bitrate_configurator.h
Moved bitrate config interface from Call class.
2018-02-21 15:03:45 +00:00
rtp_config.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_config.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_demuxer_unittest.cc
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
2017-11-22 11:21:47 +00:00
rtp_demuxer.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtp_demuxer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_rtcp_demuxer_helper_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtp_rtcp_demuxer_helper.cc
Optional: Use nullopt and implicit construction in /call
2018-01-11 12:00:00 +00:00
rtp_rtcp_demuxer_helper.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +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
Removing AvailableBandwidth method on transport controller.
2018-03-12 15:53:49 +00:00
rtp_transport_controller_send.cc
Fixes to posting delayed process tasks in SSCC.
2018-03-16 15:28:21 +00:00
rtp_transport_controller_send.h
Fixes to posting delayed process tasks in SSCC.
2018-03-16 15:28:21 +00:00
rtx_receive_stream_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
rtx_receive_stream.cc
Stop using LOG macros in favor of RTC_ prefixed macros.
2017-11-09 11:56:32 +00:00
rtx_receive_stream.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
ssrc_binding_observer.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
syncable.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
video_config.cc
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."
2018-03-01 18:07:29 +00:00
video_config.h
Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers."
2018-03-01 18:07:29 +00:00
video_receive_stream.cc
Delete member VideoReceiveStream::Config::Rtp::ulpfec.
2017-09-26 09:49:21 +00:00
video_receive_stream.h
Move videosinkinterface.h to common_video to solve a circular dep.
2018-01-04 13:19:49 +00:00
video_send_stream.cc
Make RTCP report interval configurable
2018-02-01 10:12:11 +00:00
video_send_stream.h
Reland "Add hugeFramesSent GetStats metric"
2018-03-06 13:38:11 +00:00