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Andrei Dumitru 0987273e1d Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video. 
R=nisse@webrtc.org

Change-Id: I09d03af461d7fbe200098fe91845f7b76fab6c4f

Bug: webrtc:10954
Change-Id: I09d03af461d7fbe200098fe91845f7b76fab6c4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150863
Commit-Queue: Andrei Dumitru <andreidumitru@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29114}
2019-09-09 15:39:23 +00:00
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test
Add RtpPacketPacer interface for pacer control
2019-07-29 15:37:39 +00:00
audio_receive_stream.cc
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audio_receive_stream.h
Split out RtpSource from libjingle_peerconnection_api
2019-09-02 14:04:47 +00:00
audio_send_stream.cc
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audio_send_stream.h
Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo
2019-08-22 07:23:04 +00:00
audio_state.cc
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audio_state.h
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bitrate_allocator_unittest.cc
Wire the stable target bitrate from GoogCC to the BitrateAllocator
2019-08-22 15:25:15 +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_estimator_tests.cc
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BUILD.gn
Split out RtpSource from libjingle_peerconnection_api
2019-09-02 14:04:47 +00:00
call_config.cc
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call_config.h
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call_factory.cc
DegradedCall: fake network using TaskQueue instead of ProcessThread
2019-08-06 15:05:30 +00:00
call_factory.h
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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
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
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
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fake_network_pipe_unittest.cc
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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
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
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flexfec_receive_stream_unittest.cc
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flexfec_receive_stream.cc
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flexfec_receive_stream.h
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OWNERS
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packet_receiver.h
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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_unittest.cc
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receive_time_calculator.cc
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receive_time_calculator.h
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rtcp_demuxer_unittest.cc
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rtcp_demuxer.cc
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rtcp_demuxer.h
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rtcp_packet_sink_interface.h
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rtp_bitrate_configurator_unittest.cc
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rtp_bitrate_configurator.cc
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rtp_bitrate_configurator.h
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rtp_config.cc
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rtp_config.h
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rtp_demuxer_unittest.cc
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rtp_demuxer.cc
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rtp_demuxer.h
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rtp_packet_sink_interface.h
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rtp_payload_params_unittest.cc
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rtp_payload_params.cc
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rtp_payload_params.h
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rtp_rtcp_demuxer_helper_unittest.cc
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rtp_rtcp_demuxer_helper.cc
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rtp_rtcp_demuxer_helper.h
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rtp_stream_receiver_controller_interface.h
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rtp_stream_receiver_controller.cc
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rtp_stream_receiver_controller.h
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rtp_transport_controller_send_interface.h
Delete deprecated rtc_event_log header
2019-08-07 10:58:17 +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_video_sender_interface.h
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rtp_video_sender_unittest.cc
Disable the most flaky tests on iOS.
2019-08-14 15:42:11 +00:00
rtp_video_sender.cc
Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video.
2019-09-09 15:39:23 +00:00
rtp_video_sender.h
Delete deprecated rtc_event_log header
2019-08-07 10:58:17 +00:00
rtx_receive_stream_unittest.cc
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rtx_receive_stream.cc
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rtx_receive_stream.h
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simulated_network_unittest.cc
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simulated_network.cc
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simulated_network.h
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simulated_packet_receiver.h
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ssrc_binding_observer.h
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syncable.cc
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syncable.h
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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
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video_send_stream.h
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00