- 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}
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}
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}
Integration with datagram transport will come in next CLs.
NOTE that since we now have implemented negotiation for media transport, we can replace configuration flags with field trials, but it will be done later for both media and datagram transports.
Bug: webrtc:9719
Change-Id: Icf062d030899d53d5646977ba195d1634050704b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137820
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27996}
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> 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@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
This reverts commit df5731e44d510e9f23a35b77e9e102eb41919bf4.
Reason for revert: Breaks WebRTC in Chrome FYI for all platforms.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/2966
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> 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@{#27974}
TBR=steveanton@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
# Passing all bots except for flaky webrtc_perf_tests
NOTRY=True
Bug: webrtc:10129
Change-Id: If97317f7a01b34465685fcebbeea0d7576ed7328
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137431
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27988}
Bug: webrtc:9719
Change-Id: I6e756d925917f032aa94a221706cd4241085b2a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137340
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@{#27980}
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().
Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27979}
This is a useful tool to use for unittests of code that uses
TransportFeedback as input.
Bug: webrtc:10498
Change-Id: I171b22841eb9e16a5d5b785ff45ae9df5a6ccd7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137423
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27975}
This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
event if needed and exposes the method on RtpSenderInterface.
This is a spec-compliance change.
Bug: webrtc:10129
Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
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@{#27974}
This change introduces experimental datagram_transport interface and congestion_control interfaces. The goal is to integrate support for datagram transport in DTLS transport and set it up in a similar way we currently setup media_transport. Datagram transport will be injected in peer connection factory the same way media_transport is injected (we might even keep using the same factory which creates both media and datagram transports for now until we decided what to do next).
Bug: webrtc:9719
Change-Id: I80e70ce8d3827664ac5f5f7e55b706fe2dd2fbef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136782
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27943}
This also introduces an option in CreateOfferOptions for
getting the non-spec behavior (2013 vintage) back.
Bug: chromium:962860
Change-Id: I72267408a61d6eb03e9895fe38b4cc803d8cbbaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136809
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27941}
This CL removes two deprecated methods from the VideoEncoder interface:
* int32_t SetRates(uint32_t, uint32_t);
* int32_t SetRateAllocation(const VideoBitrateAllocation&, uint32_t);
These are no longer used, instead the new version must be implemented:
void SetRates(const RateControlParameters&) = 0;
Migrating is straight forward. For the old SetRates, simple replace:
int32_t MyEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
with
void MyEncoder::SetRates(const RateControlParameters& parameters) {
uint32_t bitrate = parameters.bitrate.get_sum_kbps();
uint32_t framerate =
static_cast<uint32_t>(parameters.framerate_fps + 0.5);
For SetRateAllocation, replace:
int32_t MyEncoder::SetRateAllocation(
const VideoBitrateAllocation& allocation,
uint32_t framerate) {
with
void MyEncoder::SetRates(const RateControlParameters& parameters) {
const VideoBitrateAllocation& allocation = parameters.bitrate;
uint32_t framerate =
static_cast<uint32_t>(parameters.framerate_fps + 0.5);
Two more things to note:
1. The new method is void. Previously the only use of the return value
in production code was to log a more or less generic error message.
Instead, log the actual error from the encoder when it happens,
then just return.
2. The new method is pure virtual; it must be overriden even in test.
This CL is intended to be landed two weeks after creation, on Thursday
May 9th 2019.
Bug: webrtc:10481
Change-Id: I61349571a280bd40cd100ca9f93c4aa7748ed30d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134214
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27926}
Intended to ease transition to new log format.
Bug: webrtc:6463, webrtc:8111
Change-Id: Icadaedb6a6a7d31038a45ff5eb0b054528f00f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135944
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27920}
Recommended usage is to create copies with Clone() and to
call JsepSessionDescription::Initialize using std::move.
Bug: webrtc:10612
Change-Id: I626a08a35ba8e112471cec0374c944e96f8effbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135941
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27911}
This CL removes the usage of absl::flat_hash_map because it transitively
depends on CCTZ which fails to link with lld-link after the switch to
libc++.
Since std::map doesn't support heterogeneous lookup until C++14, this
CL also stops using absl::string_view and switches to
`const std::string&`.
Bug: webrtc:10605
Change-Id: I4fc93969c6fc0cc7e7e62b4d2f801bdd27cff0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135566
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27877}
Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL
adds these to the GetStats API.
Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27839}
Make Vp8FrameBufferController::UpdateConfiguration return a set
of desired overrides. These overrides are cumulative with
previously returned override sets.
Bug: webrtc:10382
Change-Id: I1aa9544ae0cf6c57115e80963b3bbcdc3101db5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134649
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27835}
Add method to get real test execution time, where test execution time is
time from call setup to call terminated.
Bug: webrtc:10138
Change-Id: I7ae3995c0051ecb4fc796b895be1180c8aab77cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134302
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27822}
SetCodecPreferences allows clients to filter and reorder codecs in their
SDP offer and answer.
Bug: webrtc:9777
Change-Id: I716bed9b06496629b45210883b286f599c875239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129727
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27817}
The metrics are now added as RTCNonStandardStatsMember objects in
RTCMediaStreamTrackStats. Unit tests are updated.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/134303,
with fixes.
TBR=kwiberg@webrtc.org
Bug: webrtc:10549
Change-Id: I29dcc6fbfc69156715664e71acfa054c1b2d9038
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134500
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27806}
In this CL:
- Assign frame IDs so that simulcast streams share one frame ID space.
- Added a CodecBufferUsage class that represent how a particular buffer
was used (updated, referenced or both).
- Calculate frame dependencies based on the CodecBufferUsage information.
Bug: webrtc:10342
Change-Id: I4ed5ad703f9376a7d995c04bb757c7d214865ddb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131287
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27784}
Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely.
Adds an ENCODER and sets it up to parse SDP config for multistream
opus.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27775}
Removing functionality to choose congestion controller implementation,
using injection instead. Also cleaning up some related functionality
that's no longer needed, such as the injection of event logs into the
factory.
Bug: webrtc:9883
Change-Id: Ia528005625430ae31a15bc88881e2d4ac6ad1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133890
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27768}
QualityMetricsReporter helps to keep network emulation framework and
peer connection level test framework separated. Also it provides
ability to gather statistics from any component around with
correlation with call start and end.
Bug: webrtc:10138
Change-Id: Ib3330a8d35481fde77fcf77d2271d6cfcf188fec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132718
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27759}
