OpenH264 cannot be usually used everywhere as it's proprietary and for
that reason it's usually disabled or apps using it are not allowed to be
available in default installations. Using system OpenH264 option allows
us to use e.g. noopenH264, that can be present in default installations
and later replaced by OpenH264 installed from 3rd party repository.
Bug: webrtc:14717
Change-Id: I015aacdb48c0636935f611459f0c9a6aa74a8f94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349301
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42509}
Allow API users to access the NetworkControllerInterface instance that a
given PC ended up with, to allow integrators who have provided a
PeerConnectionFactoryDependencies.network_controller_factory to
associate a created instance of their custom network controller with the
PC using it.
Eg for the RTCRtpTransport Chromium implementation as in crrev.com/c/5607744.
Bug: chromium:345101934
Change-Id: Ia712ca4f45b90d5078f4e8e5977622d3e9f9aa6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353980
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42506}
If the task queue is blocked, there is a risk that delay becomes negative. Therefore, use max of calculated time to next schedule and 0.
Bug: webrtc:42224804
Change-Id: Ibae9000192d5042cf62b46d93e8364b58dae0d82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354880
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42501}
Update AudioEncoderFactoryTemplate implementation to expand unit tests to the new api.
Bug: webrtc:343086059
Change-Id: Ib63640de38aa15cc36067d5a3d1de2bf42cec313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353981
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42500}
Using DeinterleavedView<> simplifies these two classes, so now the
classes are arguably thin wrappers on top of DeinterleavedView<> and
AudioFrameView<> can be replaced with DeinterleavedView<>.
The changes are:
* Make VectorFloatFrame not use a vector of vectors but rather
just hold a one dimensional vector of samples and leaves the mapping
into the buffer up to DeinterleavedView<>.
* Remove the `channel_ptrs_` vector which was required due to an
issue with AudioFrameView.
* AudioFrameView is now a wrapper over DeinterleavedView<>. The most
important change is to remove the `audio_samples_` pointer, which
pointed into an externally owned pointer array (in addition to
the array that holds the samples themselves). Now AudioFrameView
can be initialized without requiring such a long-lived array.
Bug: chromium:335805780
Change-Id: I8f3c23c0ac4b5a337f68e9161fc3a97271f4e87d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352504
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42498}
Using 1 as channel_id doesn't make it clear that the goal was to
provide an invalid channel.
Bug: webrtc:345518625
Change-Id: Ie64f25b9398eafd3d0a9c8bab106e5277adef7df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353984
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42494}
This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/354622 that breaks client code using nalus_length.
No-Try: true
Change-Id: Ic0fc41696e408adefe4eb8792150a64b1eab49da
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354840
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42493}
Some DMAbuf types don't properly implement MAP_PRIVATE as it requires
copy-on-write support. As we don't need to write to these buffers, we
can switch to MAP_SHARED instead, making it work reliably on current
kernels without having any drawbacks in this context.
Tested and confirmed with libcamera software ISP on Thinkpad X13 with
an arm processor.
Bug: webrtc:42225999
Change-Id: Ic47b8c90456cccf3742e8274945dbd64fb8aac6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354623
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42484}
Do not add offset to the "length" argument for mmap call as it should be
passed as the last argument instead. This was not causing any problems
since the offset is usually 0, but it's still better to do it correctly.
Bug: webrtc:42225999
Change-Id: If1dbe7dfd2fb22c53493c0fafd23d782f0683a11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354521
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42481}
This simplifies the following steps:
* FrameCombiner infers the sample rate from channel size
* Sends the inferred sample rate to FixedDigitalLevelEstimator
and Limiter.
* Those classes then convert the sample rate to channel size.
Along the way perform checks that the derived channel size value
is a legal value (which has already been done by FrameCombiner).
To:
* FrameCombiner sends channel size to FixedDigitalLevelEstimator and
Limiter.
Bug: chromium:335805780
Change-Id: I6d2953ba5ee99771f3ff5bf4f4a049a8a29b5577
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352581
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42480}
Before the schedule starts an absl::AnyInvocvable is executed every time
a packet is enqued. The incocable should return true, if the schedule should
be started.
The pupose is to allow tests to not start a schedule until ICE and DTLs
is connected.
Bug: webrtc:42224804
Change-Id: I61bd63508830f7c27d86f982299ce2be180ff460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354464
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42479}
This CL modifies RtpTransceiver.setCodecPreferences to return RtcError
instead of void, making it easier to handle errors when setting
codec preferences. To achieve this, new RtcException and RtcError
classes are introduced to represent errors in WebRTC,
mimicking api/rtc_error.h in C++.
Bug: webrtc:42225493
Change-Id: I0f4c6e56f8f2af3353915a41084f6b7b46d793d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352900
Reviewed-by: Zoé Lepaul <xalep@webrtc.org>
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42476}
A recent change in the link capacity parameter from int to DataRate
broke the implicit mapping of 0 kbps to infinite capacity, causing
tests to fail unless an explicit capacity was specified. This
change updates the following tests to use infinite capacity by default:
screenshare_loopback
sv_loopback
video_loopback
This fix restores the expected behavior and maintains backward
compatibility.
Bug: webrtc:42224804
Change-Id: I244ea3a0f8f83a81f2dbcf40e5ff921e326f24e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354540
Reviewed-by: Erik Språng <sprang@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42475}
Adds support for 32 bits formats needed for libcamera software ISP. This
is needed, because libcamera enforces 8 byte alignment and we only
support 3 byte alignment for RGB. This will make it work with 32 bits
aligned output formats recently added to libcamera.
Relevant libcamera patch: https://patchwork.libcamera.org/patch/20253/
This has been verified on an snapdragon device using libcamera and software ISP and on my machine using "vivid" virtual camera from libcamera and enforcing specific format.
Bug: webrtc:346808586
Change-Id: I8d89120660b2304b880d952c5acd7f5cd09b611e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354400
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42474}
keeping around the DTLS 1.0 constant for unit tests.
BUG=webrtc:40644300
Change-Id: I6d0c3ba1f434bbf3ef1a1b812aeef26943dcf646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352530
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42471}
Comment out device_status as it seems to be unreliable.
Update iOS simulator runtime because generate_buildbot_json.py was complaining.
Bug: None
Change-Id: I34c3f43ebe23597351fc91884d991ef3241ac3ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354520
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#42470}
If the field trial WebRTC-DataChannelMessageInterleaving is set, message
interleaving in SCTP (RFC8260) will be enabled in dcSCTP.
Bug: webrtc:41481008
Change-Id: I989b9ca554439ab0afd71f04d14a5cb5444b3361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354480
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42469}
Set layer QP limits equal to QP limits in the main encoder config. This reduces number of nodes to modify if you need to change the settings.
Bug: b/337757868
Change-Id: Id7f6f9d6527903e8e22ff4fad2c974bee6e87cb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353982
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42466}
Along the way slightly simplify the class interface since views
carry audio properties. Also, now allocating FrameCombiner allocates
the mixing buffer in the same allocation.
Bug: chromium:335805780
Change-Id: Id7a76b040c11064e1e4daf01a371328769162554
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42465}
This avoids having duplicate camera entries presented to the user when
PipeWire camera is being used.
Bug: webrtc:346350844
Change-Id: I423db7fe0654cc1b1c91ee5264c6ba5dc4e24100
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42462}
To make Environment available for creating AudioEncoders in follow ups
Bug: webrtc:343086059
Change-Id: I0965155915caeee28964ce8406045beeabaa0185
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353741
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42460}
