This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
Removes all remaining usage of SetType and marks the method as
deprecated.
Bug: none
Change-Id: I98dc613978ffe7ad8a4ffd951dd974d56ed92983
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265100
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37137}
This moves the construction of StunMessage instances for
ConnectionRequest, outside of the Prepare() method.
Following this, removing Construct()+Prepare() is relatively
straight forward.
Bug: none
Change-Id: Ibcf0510cef30a6e648005b43602c7ae1fb06729e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264558
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37122}
* Add ctors for providing the type and transaction id at construction.
* Update tests to use them instead of SetType+SetTransactionID
* Make sure stun message enum types are based on uint16_t
* Mark SetTransactionID as deprecated.
* Mark SetStunMagicCookie as deprecated (unused in webrtc).
* Add SetTransactionIdForTest for the one test that uses it (might not
actually need it)
* Make StunRequest::Construct() protected.
* Add a TODO to follow up on this since construction of StunRequest
goes through an unnecessarily complex 3-step process involving
other classes and a virtual method.
Bug: none
Change-Id: Ib013e58f28e7b2b4fcb3b3e1034da31dfc93e9d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264546
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37079}
Currently has the same contents as struct SpatialLayer. Intention is
to add a ScalabilityMode member, which isn't appropriate for a spatial
layer.
Bug: webrtc:11607
Change-Id: I75c9e9b39407e3f24ec117bb17dc37830076b26f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262255
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37062}
This cl/ creates a constructor for a FieldTrials object that is
not backed by the global string. Use with care!
Bug: webrtc:10335
Change-Id: I8c48d1e8bb52fef78524d890cc90473355be617f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264461
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37057}
This currently only exists as a goog legacy stat and has no spec
equivalent according to
https://docs.google.com/document/d/1z-D4SngG36WPiMuRvWeTMN7mWQXrf1XKZwVl3Nf1BIE/edit
Yet it is useful to debug issues sometimes. Exposing it as a
nonstandard stat will make it show up in chrome://webrtc-internals,
removing a need to switch to the legacy stats API there.
BUG=webrtc:14118
Change-Id: I506357ad54ff33df3ba46fb81558aa32187ac8e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37055}
Step one in making it a separate type, that will be done as a
followup, after downstream code is updated to use the new name.
Bug: webrtc:11607
Change-Id: I6fa664a0729b1cfd71b7f02b6441880beee0e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262806
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36946}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
Now only using the complexity from the main VideoCodec settings.
Bug: webrtc:13694
Change-Id: I5a29df0fac0c0686bf5ea0b677f8946d23ef9888
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262762
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36912}
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36890}
VideoSimulcastConfig::encoding_params and VideoConfig::min/max_encode_bitrate_bps
were replaced by VideoConfig::encoding_params.
All usage of the previous options has been updated to the new option.
Bug: webrtc:11607
Change-Id: I52cd9efa6e640929485da7aa1e61d15a1a693b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261949
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36865}
Similar mocks are used internally, there should only be one in WebRTC.
Bug: None
Change-Id: Ic5163ae0c554c602344a0d25d17c3f0d46fc2e95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261955
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36864}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
The webrtc::VideoStreamDecoderInterface was basically created as a public version of FrameBuffer2, but to hide the complexity of FrameBuffer2 it was also combined with decoding so that the public API could be reasonably simple to use. FrameBuffer3 has a simple API with a clear purpose, so its API can be exposed directly.
Bug: webrtc:14026
Change-Id: I81dc84b869e4d16c5e02feb5c876fbcede3d4a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261181
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36781}
The feature is now enabled in other ways. See PSA or linked Monorail issue for details.
https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
Bug: webrtc:11539
Change-Id: I0f5816baf2bfa1508a1c85ddbd7b775417434c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260107
Auto-Submit: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36742}
With the encoding parameters in the SimulcastConfig objects, it wasn't
possible to configure explicit encoding parameters when using singlecast,
required for example to use the spec standard SVC API.
Bug: webrtc:11607
Change-Id: I92b1446e772e2ecec93379dc91a3da159b8bc209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260002
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36731}
