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webrtc/call
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Anton Sukhanov 4f08faae82 Introduce MediaTransportConfig 
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
2019-05-21 18:58:33 +00:00
..
test Revert "Fix target bitrate RTCP messages behavior for SVC streams" 2019-03-29 15:52:11 +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 Introduce MediaTransportConfig 2019-05-21 18:58:33 +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 ClangTidy fixes for call/ 2019-03-14 09:38:01 +00:00
bitrate_allocator.cc Enable experiments with audio bitrate priority. 2019-05-08 14:21:01 +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 Adds route changes in event logs. 2019-05-13 10:41:40 +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 Introduce MediaTransportConfig 2019-05-21 18:58:33 +00:00
call_unittest.cc Introduce MediaTransportConfig 2019-05-21 18:58:33 +00:00
call.cc Introduce MediaTransportConfig 2019-05-21 18:58:33 +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 Makes srte owner in call/ and test/ 2018-07-13 08:39:41 +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 Adding rtcp report interval into RTCConfiguration. 2018-11-12 20:00:00 +00:00
rtp_config.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_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 Forward LossNotification from RTCPReceiver to EncoderRtcpFeedback 2019-04-09 11:13:39 +00:00
rtp_transport_controller_send.cc Adds route changes in event logs. 2019-05-13 10:41:40 +00:00
rtp_transport_controller_send.h Adds route changes in event logs. 2019-05-13 10:41:40 +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 Remove redundant feedback_packet_seq_num_set_ in RtpVideoSender 2019-05-08 15:37:00 +00:00
rtp_video_sender.cc Remove redundant feedback_packet_seq_num_set_ in RtpVideoSender 2019-05-08 15:37:00 +00:00
rtp_video_sender.h Remove redundant feedback_packet_seq_num_set_ in RtpVideoSender 2019-05-08 15:37:00 +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 Introduce MediaTransportConfig 2019-05-21 18:58:33 +00:00
video_receive_stream.h Introduce MediaTransportConfig 2019-05-21 18:58:33 +00:00
video_send_stream.cc Reland "Add more adaptation text logs for VideoSendStream." 2019-05-02 11:44:06 +00:00
video_send_stream.h Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video. 2019-05-17 18:53:20 +00:00