This is needed for Chromium. The video capture API in Chromium expects the
raw frames and it will always convert or copy the frame. With the existing
API that would mean copying the frame twice.
Bug: webrtc:13177
Change-Id: I71f6e2dc6d5a812c3641ac691b75d50178fa0de7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264548
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39095}
Also set xcode back to xcode 13 for iOS 14.
Change-Id: Ic5475d274895b5f86e4fea36805dec4486adc79b
Bug: b/264630045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290894
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39094}
The goal of this bot is to replace ios_sim_x64_dbg_ios(12, 13 and 14).
Change-Id: I6d8f5004a9440f5fd8cb96730dc2dbb4abba2e61
Bug: b/264630045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290893
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39086}
There is now only one sequence number unwrapper so this is redundant.
Bug: webrtc:13982
Change-Id: I210378e069366eb21dea0051f83d7431e4177057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290892
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39085}
PacketArrivalMap explicitly doesn't promise packet at the beginning
of it is received. Ensuring that property is wasteful
Bug: chromium:1382563
Change-Id: Ifc898b7ec2bc7a302af8dcfd233e0c598f62db95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290501
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39083}
Fallback to a default value if the scalability mode is unset or not supported by the codec.
The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set).
Bug: webrtc:11607
Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39074}
This is required by some virtual cameras, like Snap Camera from
Snapchat.
Bug: webrtc:14783
Change-Id: I3d841936c17f3f227af9a94a4c3b0f37940d43b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288361
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39073}
We found that the legacy assumption for H264 which assumed that
simulcast streams would use 2x width ratios in unnecessary as the
encoder has since been fixed to handle multiple ratios.
H264 encoder still works even if this assumption is invalid
Bug: None
Change-Id: I9caacf78d26c8215b94858a2d8674ec4cd64e96e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286940
Reviewed-by: Mirta Dvornicic <mirtad@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39072}
to remove duplicated implementation of these functions between av1 packetizer, av1 depacketizer and video allocation rtp header extension
Bug: None
Change-Id: I30049f31c289bdb9e0aad6520f5145d1f999e635
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290731
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39069}
Some of the new targets are otherwise not visible by external users.
Bug: webrtc:9838
Change-Id: I7b92803692de64e0f93c9ec582c9fe615fda5e65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290844
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39068}
in favor of the Timestamp constructor and method.
The constructor is most likely not used outside libWebRTC,
the call to
.timestamp_us()
can be replaced with
.timestamp().us()
BUG=webrtc:14813
Change-Id: Id166b4f85b2425ecec1c7ebb81406f82ff9d95c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290727
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39066}
Perf tests upload its results to CPD.
With the current design, an assertion failure in one test prevents the upload for all the tests.
https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Mac%20M1%20Arm64%2012/1719/overview
The "quick" perf test mode is made to run on regular CQ/CI bots without any metrics upload so it's fine to have an assertion failure there.
Bug: b/264502081
Change-Id: I22e8e8b7ce317f43297cb8837694e420cd80613d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290571
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#39063}
Some of the new targets are otherwise not visible by external users.
Bug: webrtc:9838
Change-Id: Idc585f45eeeb937802d5a898be57cf3d887fe142
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290730
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39059}
With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused.
Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid.
Bug: webrtc:7135, webrtc:14795
Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39058}
Adds a new StatExposureCriteria for non-standard stats. This removes the
virtual call to is_standardized() which can simply use the
StatExposureCriteria.
Bug: webrtc:14546
Change-Id: If4174019ff8cc6559ab0dc9a04e0f8a6631b9842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279045
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39057}
RtpPacket has CopyOnWriteBuffer and std::vector that can be moved more
efficiently than copied, thus move of the RtpPacket is also more efficient
Bug: None
Change-Id: I5509346e426cd32d0fb0649ef1a6883b7176df1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290726
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39053}
Unsignaled SSRCs are only applicable for the receiver case (not sender).
This CL updates the receievr's GetParameters() and GetSources() methods
to lookup parameters/sources by the current SSRC (whether or not it was
signaled) instead of only looking at the signaled SSRC.
To clarify that the `ssrc_` variable inside the [Audio/Video]RtpReceiver
is the signaled ssrc (and not set if the current ssrc is unsignaled),
we rename this variable to `signaled_ssrc_`.
By the looks of it, other APIs like setting volume or packetizers also
have a dependency on the assumptions that the SSRC is signaled. We will
not address that in this CL, but this CL makes that more clear.
Bug: webrtc:14811
Change-Id: I32c93d264ab441ade23a4078639744d25b791742
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290573
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39051}
and in DegradedCall. In DegradedCall - ThreadPacketReceiver is no longer needed.
Implementation of DeliverRtpPacket is done in preparation of https://webrtc-review.googlesource.com/c/src/+/290540, where the parsed packet will be propagated to Call without extra parsing.
Bug: webrtc:7135, webrtc:14795
Change-Id: Ic068105d6d1f337afc6b4539b0e7184e736e7ee0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290704
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39048}
