That means it does not have to be set on every update of StreamsConfig.
BUG=webrtc:9586
Change-Id: I6a348160e209042857c4475323466e2aa92adef8
Reviewed-on: https://webrtc-review.googlesource.com/c/116690
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26184}
This wires up the "state change" event and defines an observer
class that can be used by clients.
Bug: chromium:907849
Change-Id: I3cba2dc051a56280fb958f139f29cbb0022a39c6
Reviewed-on: https://webrtc-review.googlesource.com/c/114884
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26120}
Currently media transport can't log events to event log, but it should (things like bitrate estimates, goog cc logging, etc). This change make RtcEventLog available inside media transport.
Bug: webrtc:9719
Change-Id: I89a3b727049ccadc11c26c1d26ebaee3a1172556
Reviewed-on: https://webrtc-review.googlesource.com/c/115789
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26106}
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.
Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
This allows rtc::scoped_refptr to be used with templates
that use element_type as the mechanism to interface with
smart pointers.
Bug: None
Change-Id: Ie742f416a78efce0b07cfa3009d939e51506ccf9
Reviewed-on: https://webrtc-review.googlesource.com/c/115100
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26078}
This will be used to calculate a lower bound for the round trip time in
a later CL.
Bug: webrtc:9718
Change-Id: I0a1d22045961fe6bd343d1d6ce9b36490b036bb1
Reviewed-on: https://webrtc-review.googlesource.com/c/114680
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26050}
CallTest, VideoQualityTest and VideoAnalyzer used test::TestVideoCapturer
as an interface for video sources. Change to use VideoSourceInterface instead,
since that's all they need.
This is a preparation for making test::VcmCapturer usable as a
VideoTrackSource, and replace use of cricket::VideoCapturer in example code.
Bug: webrtc:6353
Change-Id: I445f5f6f9b7342230b89f53a5722df9c9e92834f
Reviewed-on: https://webrtc-review.googlesource.com/c/114881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26047}
The class has been renamed to EncoderSimulcastProxy.
Bug: webrtc:10069
Change-Id: Ief03cfb27145798ac46692d9e51371d2e119eeb0
Reviewed-on: https://webrtc-review.googlesource.com/c/114551
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26031}
These metrics by themselves won't be as useful, unless they can be correlated to the use of the
feature 'WebRtcHideLocalIpsWithMdns'. This can be done by running a finch experiment where we turn
the feature on for a % of users, we can then compare these metrics for users with and without
the feature turned on.
A complementary change is required in Chrome:
tools/metrics/histograms/enums.xml
Bug: webrtc:9605 webrtc:10091 chromium:914452
Change-Id: Ibc6d16dec95a8e3943ce40063c02903769fe1cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/113321
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26003}
The address and the related address of local candidates are sanitized
accordingly when the mDNS concealment of local IPs is enabled. Also,
remote hostname candidates created from signaling are sanitized in stats
as well. A couple of unit tests are revised to reflect the desired
behavior of AsyncResolverInterface so that when a hostname candidate is
resolved, the hostname is kept in the candidate address.
Bug: webrtc:9605, chromium:914452
Change-Id: Iad9ad04ce4e50304e44cf04b15b97a7ae2dec960
Reviewed-on: https://webrtc-review.googlesource.com/c/113643
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25996}
- Add chroma siting to color space RTP extension.
- Use 16 bits for max/min luminance.
- Change denominator of chromaticity and luminance.
- Add RTC_DCHECKs to protect against overflows.
Bug: webrtc:8651
Change-Id: If8b95bad6241381224eaba9c5bccce06a65a9195
Reviewed-on: https://webrtc-review.googlesource.com/c/113804
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25990}
Bug: None
Change-Id: I0fd0616132705c6d15a77fc442be47080f1b81b1
Reviewed-on: https://webrtc-review.googlesource.com/c/112721
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25975}
The flag rtc_use_builtin_sw_codecs will be removed in a later CL and
this marks usage of the various entry points using the old video factory
API as deprecated.
Bug: webrtc:7925, webrtc:10044
Change-Id: I5c75516a41b0666e77539c028808cc2b173ed4bd
Reviewed-on: https://webrtc-review.googlesource.com/c/113061
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25970}
Also adds a set_buffer method, as the only public setter for capacity_.
Bug: webrtc:9378
Change-Id: If0257c6d00bc8690f0428a3edc20b6da6dfa7119
Reviewed-on: https://webrtc-review.googlesource.com/c/112134
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25962}
This change introduces a new measurement into the VideoReceiveStream::Stats
structure to measure the latency between the first frame being received and
the first frame being decoded in WebRTC. The goal here is to measure the latency
difference when a FrameEncryptor is attached and not attached.
Change-Id: I0f0178aff73b66f25dbc6617098033e226da2958
Bug: webrtc:10105
Reviewed-on: https://webrtc-review.googlesource.com/c/113328
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25956}
This reverts commit cdc5eb0de1.
Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().
Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
>
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
>
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}
TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
Since a lot of native users have taken dependencies on our old, non-standard behaviour
we'll have to have two ice connection states living side by side until we can get rid
of the old one.
Bug: webrtc:6145
Change-Id: I9b673bffeb1dfcf410f7c56d4def5912121e644c
Reviewed-on: https://webrtc-review.googlesource.com/c/113421
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25929}
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.
Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
Use the new class internally where appropriate too.
The objective is to rename it, but due to some external dependency,
it is better to copy, update dependencies and remove.
Bug: webrtc:10069
Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3
Reviewed-on: https://webrtc-review.googlesource.com/c/113265
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25920}
Replaces enum VideoCodecType for video frames and uint8_t for audio
frames.
Also delete method
MediaTransportVideoSinkInterface::OnKeyFrameRequested; it needs to be
added as a send-side interface instead (for a later cl).
Bug: webrtc:9719
Change-Id: I2cfdbacc267afc75c448512e2cc6de0ec9966a2d
Reviewed-on: https://webrtc-review.googlesource.com/c/113180
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25918}
Prior to this CL, if the "a=msid" attribute was missing it was treated
the same as if "no streams" were explicitly signaled (a=msid:-); the
receivers would not be associated with any streams.
In order to support legacy endpoints that don't recognize "a=msid" that
assume the Plan B behavior of a stream being created anyway, this CL
creates a stream with a random ID in such cases. For background, see
https://github.com/web-platform-tests/wpt/pull/14054.
Bug: chromium:907508
Change-Id: I9d9dd0e4ba8f9941f8652f4d7873adc560777cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/112900
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25901}
- Enable vp9 flexible mode in VideoEngine if 3 spatial layers are set.
- Enable flexible mode in loopback tools and quality tests.
- Reset first active spatial layer on keyframe in encoder.
- Ensure duplicate references are not set by the sender in video header.
- Set references manually for flexible mode in vp9 encoder.
- Delay new activated layers until next base layer frame.
- On receive side put each spatial layer as a separate frame to FrameBuffer
and return several frames combined from FrameBuffer.
Bug: webrtc:10049,webrtc:9794,webrtc:9784
Change-Id: I01e69f134cc145deba666ccc92deb1d37a324ede
Reviewed-on: https://webrtc-review.googlesource.com/c/112289
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25895}
This setter method is intended to replace all direct assignments to
the |_size| member. A later cl will make that member private, and
rename it to |capacity_|.
Bug: webrtc:9378
Change-Id: I37e9eb54d1c72bcd4cb8a1cfef34bbc6c209bd0d
Reviewed-on: https://webrtc-review.googlesource.com/c/113060
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25894}
