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Piotr (Peter) Slatala 179a3923b9 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. 
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.


Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
2018-11-16 19:31:11 +00:00
..
test
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
audiotrack.cc
Remove "using rtc::scoped_ptr" from audiotrack.cc.
2017-11-28 21:29:18 +00:00
audiotrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
BUILD.gn
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
bundlefilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.h
Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""
2018-03-29 02:45:17 +00:00
channel_unittest.cc
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
2018-10-30 08:06:49 +00:00
channel.cc
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
2018-11-16 19:31:11 +00:00
channel.h
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
2018-11-16 19:31:11 +00:00
channelinterface.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
channelmanager_unittest.cc
Reland "Reland "Propagate media transport to media channel.""
2018-10-17 20:54:06 +00:00
channelmanager.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
channelmanager.h
Reland "Reland "Propagate media transport to media channel.""
2018-10-17 20:54:06 +00:00
createpeerconnectionfactory.cc
Revert "Use the factory instead of using the builtin code path in VideoCodecInitializer."
2018-11-01 04:46:02 +00:00
datachannel_unittest.cc
Removing warning suppression flags from pc/.
2018-07-04 10:35:27 +00:00
datachannel.cc
Implement data channels over media transport.
2018-11-09 00:40:32 +00:00
datachannel.h
Implement data channels over media transport.
2018-11-09 00:40:32 +00:00
DEPS
Allows injection of network controller factory into peer connection factory.
2018-05-18 17:07:16 +00:00
dtlssrtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
dtlssrtptransport.cc
Compute RTCConnectionState and RTCIceConnectionState.
2018-10-22 11:33:17 +00:00
dtlssrtptransport.h
Compute RTCConnectionState and RTCIceConnectionState.
2018-10-22 11:33:17 +00:00
dtmfsender_unittest.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.h
Use AsyncInvoker in DtmfSender instead of MessageHandler
2018-08-31 22:49:26 +00:00
externalhmac.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
externalhmac.h
[CFI] Fix cfi-icall failures
2017-12-12 09:27:07 +00:00
iceserverparsing_unittest.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
jsepicecandidate.cc
Delete unused utf8 conversion utilities
2018-10-11 16:16:33 +00:00
jsepsessiondescription_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
jsepsessiondescription.cc
Move SdpType from/to string definition close to declaration.
2018-10-12 09:59:40 +00:00
jseptransport_unittest.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
jseptransport.cc
Plug-in media transport state listener
2018-11-01 15:52:56 +00:00
jseptransport.h
Plug-in media transport state listener
2018-11-01 15:52:56 +00:00
jseptransportcontroller_unittest.cc
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
jseptransportcontroller.cc
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
jseptransportcontroller.h
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
localaudiosource_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
localaudiosource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
localaudiosource.h
AudioSource allows implementations to return settings
2018-11-13 16:30:09 +00:00
mediaconstraintsinterface_unittest.cc
Rename a few MediaConfig::Video flags for consistency.
2018-01-22 17:32:58 +00:00
mediasession_unittest.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
mediasession.cc
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
mediasession.h
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
mediastream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamobserver.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
mediastreamobserver.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
mediastreamtrack.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Adding shampson (me) as an owner to pc/ & api/.
2018-07-03 20:39:17 +00:00
peerconnection_bundle_unittest.cc
Add CreateIceCandidate overload which takes a cricket::Candidate
2018-07-23 23:49:44 +00:00
peerconnection_crypto_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection_datachannel_unittest.cc
Implement data channels over media transport.
2018-11-09 00:40:32 +00:00
peerconnection_histogram_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection_ice_unittest.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
peerconnection_integrationtest.cc
Fix up an outdated comment in peerconnection_integrationtest.cc.
2018-11-09 18:53:05 +00:00
peerconnection_jsep_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection_media_unittest.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
peerconnection_rampup_tests.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnection_rtp_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection_signaling_unittest.cc
Properly setup MockPeerConnectionObserver in tests (continued).
2018-11-01 09:35:07 +00:00
peerconnection.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
peerconnection.h
Pass the media transport from JsepTransportController to Call.
2018-11-15 17:36:48 +00:00
peerconnectionendtoend_unittest.cc
Run some peer connection end-to-end tests with an empty audio encoder factory
2018-11-13 16:14:50 +00:00
peerconnectionfactory_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnectionfactory.cc
Exposing audio and video engines directly.
2018-11-16 15:40:45 +00:00
peerconnectionfactory.h
Pass MediaTransportFactory to PeerConnectionFactory.
2018-10-08 18:11:06 +00:00
peerconnectioninterface_unittest.cc
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
peerconnectioninternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
peerconnectionwrapper.h
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
proxy_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remoteaudiosource.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
remoteaudiosource.h
Configure media flow correctly with Unified Plan
2018-01-18 19:01:38 +00:00
rtcpmuxfilter_unittest.cc
Delete unused code from media/base/testutils.{cc,h}
2018-10-18 16:06:33 +00:00
rtcpmuxfilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
rtcstats_integrationtest.cc
Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
2018-11-16 17:13:40 +00:00
rtcstatscollector_unittest.cc
Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain()
2018-10-25 15:52:06 +00:00
rtcstatscollector.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtcstatscollector.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
rtcstatstraversal_unittest.cc
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtcstatstraversal.cc
RTCStatsCollector::GetStatsReport() with optional selector argument.
2018-03-19 15:32:16 +00:00
rtcstatstraversal.h
Stats traversal algorithm added.
2018-03-12 10:54:09 +00:00
rtpmediautils_unittest.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.cc
Implement legacy offer_to_receive options for Unified Plan
2018-01-27 02:20:29 +00:00
rtpmediautils.h
Reland "Reland "Remove our stream << overloads from non-test build targets.""
2018-05-03 10:41:41 +00:00
rtpparametersconversion_unittest.cc
Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
2018-11-07 15:35:56 +00:00
rtpparametersconversion.cc
Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
2018-11-07 15:35:56 +00:00
rtpparametersconversion.h
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpreceiver.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpreceiver.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpsender.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpsender.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtpsenderreceiver_unittest.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransceiver_unittest.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransceiver.cc
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransceiver.h
Prevent channels being set on stopped transceiver.
2018-11-14 16:23:07 +00:00
rtptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtptransport.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtptransportinternal.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
rtptransportinternaladapter.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
rtptransporttestutil.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
sctputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
sdputils.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
sdputils.h
Move sessiondescription.h/cc from p2p/base to pc/
2017-12-20 00:21:52 +00:00
sessiondescription_unittest.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
sessiondescription.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
sessiondescription.h
Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 12:35:45 +00:00
srtpfilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
srtpfilter.cc
Delete unused utf8 conversion utilities
2018-10-11 16:16:33 +00:00
srtpfilter.h
When SDES is used, pass pre-shared key to media transport.
2018-10-31 16:04:16 +00:00
srtpsession_unittest.cc
Remove deprecated field_trial_default and metrics_default.
2018-09-28 07:21:07 +00:00
srtpsession.cc
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtpsession.h
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
2018-10-09 12:10:55 +00:00
srtptestutil.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
srtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
srtptransport.cc
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
srtptransport.h
Delete struct rtc::PacketTime.
2018-11-05 16:21:39 +00:00
statscollector_unittest.cc
Remove deprecated AudioProcessing::GetStatistics function
2018-11-01 11:21:15 +00:00
statscollector.cc
Refactor: Renaming ssl_cert_chain to GetSSLCertificateChain()
2018-10-25 15:52:06 +00:00
statscollector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamcollection.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
trackmediainfomap_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
transportstats.cc
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
transportstats.h
Create the JsepTransportController and JsepTransport2.
2018-02-23 00:13:45 +00:00
videocapturertracksource_unittest.cc
Add default constructor for rtc::Event
2018-11-07 08:57:50 +00:00
videocapturertracksource.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
videotrack_unittest.cc
Refactor VideoTrackSource, without raw pointer injection.
2018-05-23 15:42:10 +00:00
videotrack.cc
Enable cpplint in pc/
2017-10-30 18:08:29 +00:00
videotrack.h
Reland "[cleanup] Remove useless includes."
2018-10-08 07:44:19 +00:00
videotracksource.cc
Delete deprecated VideoTrackSource constructor.
2018-06-11 19:15:10 +00:00
videotracksource.h
Export symbols needed by the Chromium component build (part 2).
2018-10-15 19:52:31 +00:00
webrtcsdp_unittest.cc
Rename too long variable name to extmap_allow_mixed
2018-10-23 09:25:19 +00:00
webrtcsdp.cc
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase"
2018-10-25 14:13:44 +00:00
webrtcsdp.h
Export symbols needed by the Chromium component build (part 7).
2018-10-25 11:41:16 +00:00
webrtcsessiondescriptionfactory.cc
Remove obsolete comment (WebRtcSessionDescriptionFactory ctor)
2018-11-12 15:24:58 +00:00
webrtcsessiondescriptionfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00