This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.
The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.
Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
The former was unused, the latter is replaced with the explicit C++11
deletions. The related RTC_DISALLOW_COPY_AND_ASSIGN is left for now,
it is used in a lot more places.
Bug: None
Change-Id: I49503e7f2b9ff43c6285f8695833479bbc18c380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185500
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32224}
Add a get_and_clear_legacy_stats flag to AudioReceiveStream::GetStats,
to distinguish calls from standard GetStats and legacy GetStats.
Add const method NetEq::CurrentNetworkStatistics to get current
values of stateless NetEq stats. Standard GetStats will then call this
method instead of NetEq::NetworkStatistics.
Bug: webrtc:11622
Change-Id: I3833a246a9e39b18c99657a738da22c6e2bd5f5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183600
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32092}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
Setting a minimum delay can fail in some cases. It is important that the
AV sync code is aware of failures and can act accordingly to recover and
prevent sync delays that keep increasing indefinitely.
Bug: webrtc:11805
Change-Id: I0deed951dc6c6d0905536a949af875e0a6d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32062}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
Potential deadlock fixed by acquiring lock before calling encoder.
This is a reland of a135557b3c7ffa4fb1710d2d907c3cb86c5d5913
Original change's description:
> Call OnReceivedOverhead after audio network adaptor is created.
>
> This prevents ending up in a state where audio network adaptor never
> receives the current packet overhead and therefore doesn't work.
>
> Bug: chromium:1086942
> Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31951}
Bug: chromium:1086942
Change-Id: I514e523c6607cee0099b87919f0f77ebec966ddd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181888
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31971}
This reverts commit a135557b3c7ffa4fb1710d2d907c3cb86c5d5913.
Reason for revert: Suspected downstream breakage
Original change's description:
> Call OnReceivedOverhead after audio network adaptor is created.
>
> This prevents ending up in a state where audio network adaptor never
> receives the current packet overhead and therefore doesn't work.
>
> Bug: chromium:1086942
> Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31951}
TBR=peah@webrtc.org,sprang@webrtc.org,jakobi@webrtc.org
Change-Id: I96a92f82f0431457d649cc7feb253f0e026eeada
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1086942
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181885
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31954}
This prevents ending up in a state where audio network adaptor never
receives the current packet overhead and therefore doesn't work.
Bug: chromium:1086942
Change-Id: I8ee2ffbb7741b342b3ec93fc89f2859a146f4ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181583
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31951}
Added error checking in AudioIngress and AudioEgress to detect situations where codecs have not been set; added additional unit tests for VoipCore
Bug: webrtc:11251
Change-Id: Ibd57e518892c76e2865b844ba866e380a565dd6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180229
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31874}
Add TimeController to the CreatePeerConnectionE2EQualityTestFixture
method as a first step to make PC level framework compatible with
TimeController abstraction.
Bug: webrtc:11743
Change-Id: I69305abc880059bf9fe1d4f2e3b7c10cf35417db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178485
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31607}
The lock in TransportFeedbackProxy could cause a dead-lock if audio is
included in transport feedback messages, and necessitated a revert:
https://webrtc-review.googlesource.com/c/src/+/178100
This CL removes that lock and in fact the entire TransportFeedbackProxy
class, and instead sets the observer at construction time.
We therefore don't need to guard the observer pointer anymore.
For further context, see also internal bug b/153893626
Bug: webrtc:10809
Change-Id: I79b08d8d0e587a59736b383c3596a26836c33d2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178207
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31583}
negotiates the RED codec for opus audio behind a field trial
WebRTC-Audio-Redundancy
This adds the following line to the SDP:
a=rtpmap:someid RED/48000/2
To test start Chrome with
--force-fieldtrials=WebRTC-Audio-Red-For-Opus/Enabled
BUG=webrtc:11640
Change-Id: I8fa9fb07d03db5f90cdb08765baaa03d3d0458cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176372
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31562}
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}
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 interface has a couple of issues. Primarily for me, it makes it
difficult work with the paced sender as we need to either temporarily
release a lock or force a thread-handover in order to avoid a cyclic
lock order.
For video in particular, its behavior is also falky since header sizes
can vary not only form frame to frame, but from packet to packet within
a frame (e.g. TimingInfo extension is only on the last packet, if set).
On bitrate allocation, the last reported value is picked, leading to
timing issues affecting the bitrate set.
This CL removes the callback interface and instead we simply poll the
RTP module for a packet overhead. This consists of an expected overhead
based on which non-volatile header extensions are registered (so for
instance AbsoluteCaptureTime is disregarded since it's only populated
once per second). The overhead estimation is a little less accurate but
instead simpler and deterministic.
Bug: webrtc:10809
Change-Id: I2c3d3fcca6ad35704c4c1b6b9e0a39227aada1ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173704
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31185}
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}
Reset the frame transformer delegate's callback to
ChannelReceive::OnReceivedPayloadData when the channel is destroyed, to
prevent future callbacks from the delegate.
Bug: chromium:870644
Change-Id: Iaa2c1b7b26dc38709d3fe64a180ccc6a60a1ec9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173751
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31121}
This is based on channel_receive.cc implementation where non-audio
related logics are trimmed off for smaller footprint in size.
Bug: webrtc:11251
Change-Id: I743c9f93f509fa6fcc12981fa73a6f01ce38348c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172821
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31117}
Since the flag is now on by default, we can remove it (after all
callers stop passing it).
We can also remove all Chart JSON code from WebRTC since it is
no longer used.
Requires one recipe CL and one downstream CL to land first.
Bug: chromium:1029452
Change-Id: Ic1d62e8ab9dfcd255cd2bf51d153db80d59c564b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171878
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30927}
The proto code is copied into the out dir, so always use that since
it is what isolate is using. Previously we pointed straight at the
checkout code.
I think copying python into the out dir is probably the right way
to do things, so we should go that way in the future.
Bug: chromium:1029452
Change-Id: I701cc84a674021d2f78c73db8808f55cd6ae5174
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171877
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30923}
For simplicity and flexibility on audio only API, it deemed
to be better to trim off all audio unrelated logic to serve
the purpose.
Bug: webrtc:11251
Change-Id: I40e3eba2714c171f7c98b158303a7b3f744ceb78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169462
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30922}
Turns out my new protobuf internal entries are copied to the out dir,
so we need a new entry for that.
Tbr: mbonadei@webrtc.org
Bug: chromium:1029452
Change-Id: I5bcae3a7ff1163e051382ae741646f206ccc7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171869
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30918}
