- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.
TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.
Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}
Prior to this CL, only the mline index of an ice candidate was used to
look up contents. However, due to recent changes, it is possible that
no mline index is specified, but that only a mid is specified.
No mline index is indicated with a -1 value.
This CL makes sure the mid is used if no mline index is given.
Bug: chromium:965483
Change-Id: I8962e71acb386f7b50349802f27358ba24c11921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138075
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28045}
This reverts commit 74b373f04a69b279e45b0792d86c594cb33d22c1.
Reason for revert: This breaks chromium, blocking webrtc from rolling.
...
In file included from ../../third_party/webrtc\rtc_base/strings/string_builder.h:23:
../../third_party/webrtc\rtc_base/string_utils.h(54,28): error: implicit conversion loses integer precision: 'std::__1::basic_string<wchar_t, std::__1::char_traits<wchar_t>, std::__1::allocator<wchar_t> >::size_type' (aka 'unsigned long long') to 'int' [-Werror,-Wshorten-64-to-32]
ws.size());
See https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8912652299012991936/+/steps/compile__with_patch_/0/stdout
Original change's description:
> Delete STACK_ARRAY macro, and use of alloca
>
> Refactor the few uses of STACK_ARRAY to avoid an extra copy
> on the stack.
>
> Bug: webrtc:6424
> Change-Id: I5c8f3c0381523db0ead31207d949df9a286c76ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137806
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28038}
TBR=kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I223fceab60855dde363cc9ce8375e8f5cca43c02
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138209
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28043}
The following private methods needlessly took a reference to the
RtpConfig on which they had worked, which was itself a member.
* ConfigureProtection
* ConfigureSsrcs
* ConfigureRids
Bug: None
Change-Id: I013ca438915336d1b8f3477fe8b9f1bf87f514f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138205
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28041}
This CL updates the video packet overhead calculation to make it more
clear. This prepares for future work on improving the accuracy of the
calculation.
Bug: webrtc:9883
Change-Id: I1d623a3e0de45be7b6e4a1f9e3cbe54fd2b8a45a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138077
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28040}
Change the way the forget factor converge to the steady state so that we don't overemphasize the first packets received.
The logic is controlled by the delay histogram field trial which has an added parameter to control if emphasis should be even (c=1, default) or put on later packets (c>1) until we reach our steady state forget factor.
Bug: webrtc:10411
Change-Id: Ia5d46c22d1a4a66994652f71c8cde664362bfacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137050
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28039}
Refactor the few uses of STACK_ARRAY to avoid an extra copy
on the stack.
Bug: webrtc:6424
Change-Id: I5c8f3c0381523db0ead31207d949df9a286c76ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137806
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28038}
WebRTC combines VP9 SVC spatial layer frames into superframe and passes
it to a decoder. The chromium HW VP9 decoder (wrapper) needs to know
location of each spatial layer frame in the frame buffer. To provide
decoder with such information this CL:
- Adds Set/SpatialLayerFrameSize methods to EncodedImage.
- Sets size of each spatial layer frame on superframe at assembly stage.
Bug: webrtc:10495
Change-Id: I68c3c0d668c67dfa1740e004059d860dd98f67f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28032}
The gathering of host candidates with mDNS names is asynchronous and its
completion can happen after a srflx candidate is gathered by the same
underlying socket. We have a broken check in UDPPort::CreateConnection()
that assumes the gathering of host and srflx candidates is sequential.
This CL also does minor refactoring and clean-up.
Bug: chromium:944577
Change-Id: Ic28136a9515081f40b232a22fcbf4209814ed33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138043
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28030}
This implements RTCAudioSourceStats and RTCVideoSourceStats, both
inheriting from abstract dictionary RTCMediaSourceStats:
https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
All members are implemented except for the total "frames" counter:
- trackIdentifier
- kind
- width
- height
- framesPerSecond
This means to make googFrameWidthInput, googFrameHeightInput and
googFrameRateInput obsolete.
Implemented using the same code path as the goog stats, there are
some minor bugs that should be fixed in the future, but not this CL:
1. We create media-source objects on a per-track attachment basis.
If the same track is attached multiple times this results in
multiple media-source objects, but the spec says it should be on a
per-source basis.
2. framesPerSecond is only calculated after connecting (when we have a
sender with SSRC), but if collected on a per-source basis the source
should be able to tell us the FPS whether or not we are sending it.
Bug: webrtc:10453
Change-Id: I23705a79f15075dca2536275934af1904a7f0d39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28028}
Scenario tests runs all its activities on task queues. This is not
allowed by the default event log writer, causing a DCHECK failure.
This CL makes it possible to stop the event asynchronously,
thereby avoiding the need for the DCHECK.
Bug: webrtc:10365
Change-Id: I1206982b29fd609ac85b4ce30ae9291cbec52041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136685
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28027}
This is a standardized metric:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
We estimate the target frame size in bytes from the current encoder
target bitrate and encoder framerate.
We would expect that the average bytes produced by the encoder would
over time match the average target, which is calculated by polling
getStats() twice and dividing the delta totalEncodedBytesTarget with
the delta framesEncoded. This is meant to make googTargetEncBitrate
obsolete.
Bug: webrtc:10446
Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28022}
RTP timestamp was recently added to contributing sources in the WebRTC
specification. This CL implements that change in WebRTC.
Bug: webrtc:10650
Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28020}
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}
This change adds missing GetTransportParametersOffer, which is required for datagram transport setup. We have exactly the same method in MediaTransportInterface. It's possible to add a separate interface, which will be used in both Media and Datagram transports, but I do not want to overcomplicate it now until we know more about future of media and datagram transports.
Bug: webrtc:9719
Change-Id: I8b6c9ebc9522acba75f74da2e18e4bb1eb0d1e4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137861
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28011}
When running unpack_aecdump --full, unpack RuntimeSettings into files, on the format that can be imported into Audacity.
Output one file for each RuntimeSetting present in the aecdump. If outputting several WAV files, output file for each WAV file with corresponding time stamps.
Bug: webrtc:10643
Change-Id: If147e509d36207f5f838457354e2451df65549d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137426
Commit-Queue: Fredrik Hernqvist <fhernqvist@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28007}
This involves catching another callback from usrsctp.
It also moves the definition of "connected" a little later
in the sequence: From "ready to send data" to the reception
of the SCTP_COMM_UP event.
Bug: chromium:943976
Change-Id: Ib9e1b17d0cc356f19cdfa675159b29bf1efdcb55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137435
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28004}
Since many tests rely on rtc::Thread::Current(), add an
explicit rtc::AutoThread in the main() function used by tests.
Bug: webrtc:9714
Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28000}
