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Erik Språng a9229043e3 Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly. 
This prepares for splitting RtpSenderEgress out of RTPSender.
For context, see:
https://webrtc-review.googlesource.com/c/src/+/158020

Bug: webrtc:11036
Change-Id: I6d385ba255ce23f4c6685a3737eeb243ce2ec6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158201
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29601}
2019-10-24 12:13:56 +00:00
..
test
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
audio_receive_stream.cc
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audio_receive_stream.h
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
audio_send_stream.cc
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audio_send_stream.h
Delete deprecated bytes_sent/bytes_rcvd stat values
2019-10-17 06:41:38 +00:00
audio_state.cc
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audio_state.h
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bitrate_allocator_unittest.cc
Propagating TargetRate struct to BitrateAllocator.
2019-09-19 14:03:04 +00:00
bitrate_allocator.cc
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_allocator.h
Converts const methods in BitrateAllocator to non-member functions.
2019-09-25 11:55:13 +00:00
bitrate_estimator_tests.cc
Hide deprecated SingleThreadedTaskQueueForTest behind an accessor
2019-10-22 11:57:49 +00:00
BUILD.gn
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
call_config.cc
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call_config.h
Remove api/bitrate_constraints.h.
2019-09-18 06:37:58 +00:00
call_factory.cc
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call_factory.h
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call_perf_tests.cc
Hide deprecated SingleThreadedTaskQueueForTest behind an accessor
2019-10-22 11:57:49 +00:00
call_unittest.cc
Delete unused method PacedSender::QueueSizePackets
2019-10-17 08:35:38 +00:00
call.cc
Replacing bandwidth adaptation trial with stable target in Opus encoder.
2019-09-24 16:35:02 +00:00
call.h
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degraded_call.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +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
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +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
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
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
flexfec_receive_stream.cc
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flexfec_receive_stream.h
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OWNERS
…
packet_receiver.h
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rampup_tests.cc
Hide deprecated SingleThreadedTaskQueueForTest behind an accessor
2019-10-22 11:57:49 +00:00
rampup_tests.h
Update call Rampup tests not to rely on DEPRECATED_SingleThreadedTaskQueueForTesting
2019-10-21 12:33:27 +00:00
receive_time_calculator_unittest.cc
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receive_time_calculator.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
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
Remove webrtc::MinPositive from api/.
2019-09-18 12:52:09 +00:00
rtp_bitrate_configurator.h
Remove api/bitrate_constraints.h.
2019-09-18 06:37:58 +00:00
rtp_config.cc
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rtp_config.h
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rtp_demuxer_unittest.cc
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
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
Reduce number of RTPVideoSender::SendVideo parameters
2019-10-11 10:59:21 +00:00
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
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
rtp_stream_receiver_controller.h
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rtp_transport_controller_send_interface.h
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
rtp_transport_controller_send.cc
Clarify NetworkControl interface: result of each function must be used
2019-10-21 12:35:07 +00:00
rtp_transport_controller_send.h
Using struct for bitrate allocation limits.
2019-09-18 17:03:59 +00:00
rtp_video_sender_interface.h
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +00:00
rtp_video_sender_unittest.cc
Cleanup: Propagating BitrateAllocationUpdate to RtpVideoSender
2019-10-15 14:40:48 +00:00
rtp_video_sender.cc
Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly.
2019-10-24 12:13:56 +00:00
rtp_video_sender.h
Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly.
2019-10-24 12:13:56 +00:00
rtx_receive_stream_unittest.cc
Fix handling of large packets in RtxReceiveStream
2019-10-10 08:39:46 +00:00
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
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
video_receive_stream.cc
Delete flag VideoReceiveStream::Config::Rtp::remb
2019-09-16 11:20:55 +00:00
video_receive_stream.h
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
2019-10-23 07:46:39 +00:00
video_send_stream.cc
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video_send_stream.h
Move MediaTransportInterface out of the libjingle_peerconnection_api target
2019-09-13 10:49:56 +00:00