Set 2 thread as target for 1280x720 pixel count, and then scale up
linearly from there - but cap at physical core count.
For common resolutions this results in:
1 for 360p
2 for 720p
4 for 1080p
8 for 1440p
18 for 4K
Bug: webrtc:11551
Change-Id: I666bd971eccddee096749f20d3b08eb40fe868ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177012
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31513}
The side effect of not filtering on V4L2_CAP_VIDEO_CAPTURE is that
every device is enumerated twice. Because we look up devices by name,
and the device that supports V4L2_CAP_VIDEO_CAPTURE seems to always
appear first in /dev/video, this does not seem to end up with us ever
choosing an inappropriate device. We might get away with just filtering
device names from the list, but if the order of devices ever changed in
/dev/video there could be problems.
Bug: webrtc:11641
Change-Id: I16fee4edc873838ed4643ee16a8bbc699d6bbcf5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176460
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Dan Minor <dminor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31508}
Previously, number of decoder threads for VP9 were always set to 8 but
with a cap at number of cores. This was done since we "can't know" the
resolution that will be used.
With this change, we now intialize the number of threads based on
resolution given in InitDecode(). If a resolution change happens in
flight, it requires a keyframe. We therefore parse the header from
any key frame and if it has a new resolution, we re-initialize the
decoder.
The number of threads used is based on pixel count. We set one thread
as target for 1280x720, and scale up lineraly from there. The 8-thread
cap is gone, but still limit it core count.
This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.
Bug: webrtc:11551
Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31507}
Adding field trial WebRTC-Audio-NetEqExtraDelay with a parameter value
to set the extra delay in NetEq. This overrides the
extra_output_delay_ms parameter in NetEq::Config.
Bug: b/156734419
Change-Id: Iae7d439fafa3059494249959ac13a02de63d6b7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176858
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31493}
Also removing some related code that appears to be unused.
This is a part of simplifying the RtpRtcpInterface implementation.
Bug: webrtc:11581
Change-Id: I580bfdc1b821d571cb7437d7713a49ee4de2d19a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176568
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31464}
The forked and deprecated implementation is used by the
deprecated ModuleRtpRtcpImpl implementation.
Change-Id: If67ca1181f40969791cf9c8903c0e49679c86834
Bug: webrtc:11581, webrtc:11611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176566
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31459}
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.
The mutex types supportable by webrtc::Mutex are
- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)
In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.
The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.
Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.
Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.
Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.
The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.
Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
This reverts commit 6958d2c6f0ce5267bdc4120d88680a4be9ed5e59.
Disable the test on iOS.
Bug: None
Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jerome Jiang <jianj@google.com>
Cr-Commit-Position: refs/heads/master@{#31437}
to unblock rolling new version where private function is no longer available
Bug: None
Change-Id: I9c35fede3f331f7688cc97acfbda1250b42348a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176441
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31427}
The method is being used externally to create instances
of the deprecated internal implementation.
Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.
Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}
This required for gcc-10.
Bug: None
Change-Id: Iba3ff0881005cb2ae4574e47284a1b881594de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176361
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31417}
This reverts commit 45bb717a2866c2d836b5332a24af0d09f2b30714.
Reason for revert: Use #if RTC_TRACE_EVENTS_ENABLED to avoid unused variable.
Original change's description:
> Revert "Add trace of enqueued and sent RTP packets"
>
> This reverts commit 45b9192ad981dcdc12ad4aef087fff2195bd030c.
>
> Reason for revert: When tracing is disabled, this results in a clang warning (unused variable), which results in a build error since Werror is enabled by default.
>
> Original change's description:
> > Add trace of enqueued and sent RTP packets
> >
> > This is useful in debugging the latency from a packet
> > is enqueued until it's sent.
> >
> > Bug: webrtc:11617
> > Change-Id: Ic2f194334a2e178de221df3a0838481035bb3505
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176231
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31381}
>
> TBR=sprang@webrtc.org,kron@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:11617
> Change-Id: I854c17e587c624691a0e5e3ec9fd38c2607eda84
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176380
> Commit-Queue: Casey Fischer <caseyfischer@google.com>
> Reviewed-by: Adam Nathan <adamnathan@google.com>
> Cr-Commit-Position: refs/heads/master@{#31399}
TBR=sprang@webrtc.org,yujo@chromium.org,adamnathan@google.com,kron@webrtc.org,caseyfischer@google.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:11617
Change-Id: I9de7f7ed290481a51c161a693f5b2d5df7d2eae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176367
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31407}
This reverts commit 45b9192ad981dcdc12ad4aef087fff2195bd030c.
Reason for revert: When tracing is disabled, this results in a clang warning (unused variable), which results in a build error since Werror is enabled by default.
Original change's description:
> Add trace of enqueued and sent RTP packets
>
> This is useful in debugging the latency from a packet
> is enqueued until it's sent.
>
> Bug: webrtc:11617
> Change-Id: Ic2f194334a2e178de221df3a0838481035bb3505
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176231
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31381}
TBR=sprang@webrtc.org,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11617
Change-Id: I854c17e587c624691a0e5e3ec9fd38c2607eda84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176380
Commit-Queue: Casey Fischer <caseyfischer@google.com>
Reviewed-by: Adam Nathan <adamnathan@google.com>
Cr-Commit-Position: refs/heads/master@{#31399}
worker_queue is used in many places and it can be confusing. This queue
is the send transport's worker queue. Rename to send_transport_queue to
reflect that.
Bug: none
Change-Id: I43c5c4cbddaee3dae1ff75aa38dc3ddee6585902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176362
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31396}
This CL moves webrtc::NackModule to a deprecated folder and annotates
the type with RTC_DEPRECATED.
Since the header should not be used outside of WebRTC, this CL doesn't
created a forward header.
Bug: webrtc:11611
Change-Id: I4d5899d473d78b8c7f4a6a018e2805648244b5f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176360
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31394}
