Queue with multiple heads is planned to be used in
DefaultVideoQualityAnalyzer to store stream state. Stream state contains
ordered sequence of frame ids that were send for this video stream.
When frame is received by one receiver it should be removed from state
for that receiver and kept for others.
How it is used can be found in this CL:
https://webrtc-review.googlesource.com/c/src/+/176411
Bug: webrtc:11631
Change-Id: Ic7fabf4d77131805a91f08a2ccfffc73c08d3e2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176402
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31444}
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}
This fixes a crash that could happen if substreams exist but there is
no kMedia substream yet. There was an assumption that we either had no
substreams or at least one kMedia substream, but this was not true.
The correct thing to do is to ignore substream stats that are not
associated with any kMedia substream, because we only produce
"outbound-rtp" stats objects for existing kMedia substreams.
A test is added to make sure no stats are returned. Prior to the fix,
this test would crash.
Bug: chromium:1090712
Change-Id: Ib1f8494a162542ae56bdd2df7618775a3473419b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176446
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31442}
Since dependencies on Abseil need to be statically linked in case
Chromium is built with is_component_build=true, this CL introduces a new
parameter for C++ library rtc_* templates (rtc_library, rtc_source_set
and rtc_static_library). This parameter (called "absl_deps") will result
in a dependency on the Abseil component (//third_party/abseil-cpp:absl)
when is_component_build=true or on the normal granular Abseil target
when is_component_build=false.
Bug: chromium:1046390
Change-Id: Iddca886926a7874488701bc9d79affb00cca72d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176447
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31441}
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}
According to gmock guidelines, mocks for API classes should live
in the same package which owns the API.
No-Try: True
Bug: webrtc:11642
Change-Id: Ib105a1806cc710bc4cff752b8950e981bb4bc326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176381
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31429}
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}
ResourceListener::OnResourceUsageStateMeasured() now takes
ResourceUsageState as argument, making Resource::UsageState()
superfluous.
With the existing "fire-and-forget" behavior of always clearing usage
state on reacting to a signal, there is no longer a need to call
ClearUsageState() so this too is removed. (We may want to have a
callback in the future to hint to the Resource that it is a good idea
to clear internal measurement samples, i.e. because the load of the
system is about to change, but we can revisit that when we need it.)
Moving the usage state to the callback has the benefit of getting rid
of the assumption that UsageState() has to return the same value every
time it is called in the same task.
This CL is also the final nail in the coffin for Resource needing to
know about the adaptation task queue: ResourceAdaptationProcessor's
ResourceListener now takes care of posting to the adaptation task
queue. To support this, the processor's SequenceChecker is replaced
by a TaskQueueBase pointer.
Bug: webrtc:11525, webrtc:11618
Change-Id: I2277e71cc3759c85b62465020935603f03792c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176376
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31416}
To support multiple participants video quality analyzer may need to know
peer names in advance to simplify internal structures and metrics
reporting.
Bug: webrtc:11631
Change-Id: I4ffb1554ab7f0e015b8e937b7ffddd55aba9826f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176364
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31415}
IsAdaptationUpAllowed is moved from Resource to AdaptationConstraint.
OnAdaptationApplied is moved from Resource to AdaptationListener.
In a future CL, Resource will be moved to api/, but
AdaptationConstraint and AdaptationListener will stay in call/.
The processor, encode stream and manager are updated to keep track of
both resources, constraints and listeners. Fakes and tests are updated.
After this CL, the manager's inner classes that prevent adaptation
implement AdaptationConstraint instead of Resource.
Bug: webrtc:11525
Change-Id: Ie9cd5b1ba7d8e161951e131ab8f6bd9d5cf765bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176368
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31409}
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 CL is in preparation for moving Resource to the api/ folder. It
does not move it, but makes it such that the moving CL can be a pure
move.
In order to do this, we must stop depending on rtc_base/rtc::TaskQueue
in favor of api/webrtc::TaskQueueBase.
There are also other rtc_base/ dependencies that we do not want to
expose to the api/ folder, like critical sections and thread
annotations which are not publically exposed. To get around this, we
make Resource an abstract interface and move all of the base class
functionality into a new non-api/ class: VideoStreamEncoderResource.
The Resource now has Register/UnregisterAdaptationTaskQueue() methods.
By explicitly unregistering, we can ensure validity of the pointer even
if the Resource outlives the PeerConnection. While public interface
methods are only to be called on the adaptation task queue, posting to
the task queue happens off-queue, so a |lock_| is introduced to guard
it.
Bug: webrtc:11525
Change-Id: I50b3a30960cdec9032016c779b47001c01dad32f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176320
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31402}
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}
