The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
ScalabilityMode should be validated against the currently
allowed codecs or the currently used codec.
Bug: webrtc:11607
Change-Id: Id2e6cbfad4f089de450150e1203657ed316e2f29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277403
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38433}
Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.
Reason for revert: crbug.com/1028937
Original change's description:
> Add support for RtpEncodingParameters::max_framerate
>
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}
TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.
Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.
Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.
Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
