Inserting old frames is not an error condition and should not print a warning, and given that it happens all the time it is also very spammy.
Bug: chromium:1066819
Change-Id: Iad2b5edc5e62822c02e2bb2a53d4318f957be3bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173022
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31172}
This patch adds detection of 5G to andoird_network_monitor
using the TelephonyManager.NETWORK_TYPE_NR.
It also adds
- TelephonyManager.NETWORK_TYPE_GSM as 2G
- TelephonyManager.NETWORK_TYPE_TD_SCDMA as 3G
- TelephonyManager.NETWORK_TYPE_IWLAN as 4G
note: AdapterTypeFromNetworkType still return rtc::ADAPTER_TYPE_CELLULAR
for all cellular connections (changing that is a next step).
Bug: webrtc:11473
Change-Id: If2e681e10b24f46ea0071db0cdba758a8c4e7ee2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174500
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31171}
RtpVideoSender now stores fec type and overhead instead of querying the
generator all the time. Setting of protection parameters and asking for
current bitrate is also now handled just by the VideoFecGenerator
instance, instead of going via RtpVideoSender.
Finally, adds method to query for RtpState in VideoFecGenerator
interface. This avoids an ugly cast that would have been even more
trouble after moving fec generation.
For context, see https://webrtc-review.googlesource.com/c/src/+/173708
Bug: webrtc:11340
Change-Id: Ia5e6cd919e71850c9cc5ed5a4f4417338d577162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174203
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31166}
This reverts commit 712ebbb5b73baf30f11711efdceb6f08248fac38.
There is apparently more usage in the wild than anticipated.
Bug: chromium:1068705
Change-Id: If2f3907e509570d305670206d8d3724413964208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31163}
This keeps usage of ToQueuedTask consistent and avoids callers having
to add additional boiler plate when using the safety flag.
From this:
tq->PostTask(ToQueuedTask([safety = my_safety_flag_]() {
if (!safety->alive())
return;
Foo();
});
to this:
tq->PostTask(ToQueuedTask(my_safety_flag_, []() {
Foo();
});
Bug: none
Change-Id: I205af56a64dd9839eb845321083d533140d614ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174262
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31161}
on the current thread.
This simplifies writing async tests that use TaskQueue and doesn't
require spinning up a new thread for simple things. The implementation
is currently based on rtc::Thread, which could also be useful in
some circumstances while migrating code over to TQ.
Remove PressEnterToContinue from the test_common files since
it's very specific and only used from one file.
Bug: none
Change-Id: I8b2c6c40809271a109ec17cf7e1120847645d58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31160}
It seems possible that getStats() and merging RTX/FlexFEC substream
stats into media substream stats can race with the creation or
destruction of the media substream that the RTX/FlexFEC substream is
associated with.
In other words, the DCHECK that ensures that there exists a stats object
to merge into is not always valid. Because there is no media stats
object to merge in to, and outbound-rtp stats objects only exists per
media SSRCs, the sensible thing to do is to RTC_LOG and ignore the
substream stats.
Bug: webrtc:11545
Change-Id: I4061d7190da7ab8bd33fa1fd92c9d819f35d76c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31156}
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:
- RTC_OBJC_TYPE_PREFIX:
Macro used to prepend a prefix to the API types that are exported with
RTC_OBJC_EXPORT.
Clients can patch the definition of this macro locally and build
WebRTC.framework with their own prefix in case symbol clashing is a
problem.
This macro must only be defined by changing the value in
sdk/objc/base/RTCMacros.h and not on via compiler flag to ensure
it has a unique value.
- RCT_OBJC_TYPE:
Macro used internally to reference API types. Declaring an API type
without using this macro will not include the declared type in the
set of types that will be affected by the configurable
RTC_OBJC_TYPE_PREFIX.
Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10
The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.
Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
This CL adds the field trial WebRTC-ForcePlayoutDelay with parameters
min_ms and max_ms. If both of these values are set, the playout delay
of any received packet will be overridden by the specified values.
Bug: None
Change-Id: I353282097e3ffa437dfc5affdfdf7780b09474e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174180
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31149}
This replaces references to the ResourceAdaptationProcessor with
references to its interface. This would make it possible to have
alternative implementations or inject fake/mock implementations for
testing.
The VideoStreamAdapter is still responsible for constructing the
ResourceAdaptationProcessor, but beyond construction it is agnostic
towards the implementation.
With this CL, I claim https://crbug.com/webrtc/11222 complete.
TBR=ilnik@webrtc.org
Bug: webrtc:11222
Change-Id: I6e7a73bf1d0b5e97bc694f66180a747b27ffb018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174160
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31148}
This CL adds miscellaneous unit tests for the
ResourceAdaptationProcessor, the VideoSourceRestrictions comparators and
the VideoStreamInputStateProvider.
Bug: webrtc:11172
Change-Id: If95f69644aaf2b43e3b19d5729bedef0b438c77b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174101
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@{#31147}
This makes it optional for an echo detector to report metrics through
the getStats interface.
Bug: webrtc:11539
Change-Id: I1fef93b7bf534637b69c16971d38709b3e849a08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31146}
Currently, ScopedJavaGlobalRef can only be set at creation and never
changed. This CL makes it possible to re-set these.
Bug: b/153389044
Change-Id: I6be92dae83a9f5f3d87aa44dde226b874f4ca0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31145}
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
The ResourceListenerResponse was used to make the QualityScaler
not clear QP samples and instead increase its frequency of checking for
QP under certain circumstances, see enum description:
https://webrtc.googlesource.com/src.git/+/c70b1028d47c1aee4892545190cd66e97d09cd55/call/adaptation/resource.h#33
Because the QualityScaler depends on whether and how adaptation
happened it should listen to adaptation happening.
This CL moves the logic that was previously in VideoStreamAdapter closer
to the QualityScaler: QualityScalerResource::OnAdaptationApplied().
This would allow the VideoStreamAdapter to operate on a separate task
queue in the future, with no dependencies on any stream-specific
resources that might operate on other task queues.
Bug: webrtc:11172, webrtc:11521
Change-Id: I07971a8a5fab5715f4ccb7d2c63f1b92bd47170f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173090
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@{#31143}
Implemented an analogue of the cpu_usage metrics from third_party/webrtc/video/video_analyzer.h for third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
Bug: webrtc:11496
Change-Id: Ifdc9daa3351f1df5db98beb8f7dc7156fc7c2a16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174020
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31141}
This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing
Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).
QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.
The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.
Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
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@{#31140}
This introduces a function AudioProcessingImpl::SetCreateOptionalSubmodulesForTesting to simulate the exclusion of build-optional submodules, and tests of the currently only excludable submodule.
Bug: webrtc:11292
Change-Id: If492606205c9fdc669a6dce3a8989a434aeeed1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173746
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31138}
We'll need to deprecate the previous classes due to being used externally
as an API.
Bug: webrtc:11489
Change-Id: I64de29c8adae304d0b7628e24dd0abc5be6387ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31136}
This patch adds a field trial string for the IceController
to the factory interface, the string is from the
"WebRTC-IceControllerFieldTrials" key.
This makes it possible to add new field trials
using that key as needed.
Bug: chromium:1024965
Change-Id: I50498e45da3c49b8e1d620c90c674eedc15dc16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31134}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
WebRTC filenames use underscores to separate words so the ssl roots
file is rtc_base/ssl_roots.h instead of rtc_base/sslroots.h.
Bug: chromium:978779
Change-Id: I2fa11c38a566e177775deb3d42230d956efc8ccc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31132}
This is important when we have multiple named devices connected over
USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
choose a specific input device to route from.
Bug: b/154440591
Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
Commit-Queue: Robin Lee <rgl@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31130}
Move creation of video sinks for dumping and showing rendered video on
screen into video quality analyzer injection helper to eliminate need
to search for video config in on track callback, which makes this more
reliable and reusable.
Bug: webrtc:11479
Change-Id: I6bb5409688fd39268f9f97bde4c9b0833a64396b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31128}
