Adding a version of AnalyzeReverseStream with audio parameters
described by StreamConfig. This is part of preparations for
multichannel APM in Chromium.
Bug: webrtc:10913
Change-Id: I7c4650eab8bd7fcdec970a7e4a8fa203f09bed9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157897
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29573}
Add new histograms
WebRTC.Video.DecodeTimePerFrameInMs.[codec].[resolution].[decoder]
These histograms are more explicit than the existing histogram
WebRTC.VideoDecodTimeMs, since they allow to see performance per
codec/resolution/decoder and also contain per frame statistics instead
of an average decode time.
There's a killswitch, WebRTC-DecodeTimeHistogramsKillSwitch, that can be
used to disable the histograms.
Bug: chromium:1007526
Change-Id: I9f75127b4bc5341e9f406c64ed91164564290b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157881
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29572}
this change is intentionally noop.
Goal is to minimize change that would replace the
SingleThreadedTaskQueueForTest with a regular task queue.
Bug: webrtc:10933
Change-Id: I6da768941af048de3716af13e41b8f0f1ccd4cab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157892
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29569}
Type mismatches will silently fail and skip reading a parameter
in the JSON parsing, except when parsing a size_t from a negative int.
This CL updates the parsing to silently ignore negative values provided
for size_t config parameters, instead of explicitly DCHECKing.
Tested: Ran the fuzzer on the crash test case with + without this fix.
Bug: chromium:1016139
Change-Id: I3899e81e1183aa54b708030efeb6e0006b8cd881
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157894
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29568}
The deleted members are bytes_sent, bytes_received,
and (receive-side) fraction_lost.
Followup to https://webrtc-review.googlesource.com/c/src/+/143178.
Bug: webrtc:10744, webrtc:10525
Change-Id: Ibd8b15619c78b82a9dc18abddb6e1687fae60c88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143902
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29566}
It is unclear what configuration the APM is running with.
This CL helps some of that. Note that the webrtc::Config
is still not reported.
Bug: webrtc:5298
Change-Id: Icc2cf0b20634c04d567c0951a57ba6fe7aa5c5b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157423
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29555}
Refactorings to the audio processing module has, piece by piece,
decreased the workload of the apm_helpers helpers. It has come to a
point where it seems more reliable to consolidate what little is left
into the WebRtcVoiceEngine itself.
Bug: webrtc:9878
Change-Id: I6d983ace8e7ccb1b99d95178cf72608a657c7506
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157443
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29553}
LogMessage::streams_ is a global and thus should have trivial destructor
Bug: None
Change-Id: Ie6a8029602f50b2bc5bab546ffc0365ef0954024
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157042
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29552}
Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to
move it to its own target and allow its dependant to avoid to take a
dependency rtc_base:on rtc_base_approved.
Bug: webrtc:9419
Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29548}
To avoid IP leak from the CreatePermission request, local relay
candidates must not be paired with remote mDNS candidates, per Section
3.3.2 in draft-ietf-rtcweb-mdns-ice-candidates-04.
Bug: webrtc:11038
Change-Id: I13aada79c812712b850293c7e17094dc8f77105a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157340
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Alex Drake <alexdrake@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29546}
This reverts commit 03bc15c646.
Reason for revert: Breaks Chromium's FYI win debug build
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win%20Builder%20(dbg)/4341
Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: Ib6032c6f8bfb6a0b5eef559430cfae4eda3da0e0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157426
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29544}
The estimator will simply compute the worst value of all combinations
of render and capture signal.
This has the drawback that low-volume or silent render channels may
severely misestimate the ERL.
The changes have been shown to be bitexact over a large dataset.
Bug: webrtc:10913
Change-Id: Id53c3ab81646ac0fab303edafc5e38892d285d8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157308
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29542}
The minimum bitrate was lower bounded by the previous value and could thus not become lower when a stream allocation was removed.
Bug: None
Change-Id: I60068dbc7691121f001cbb233ca4a25269047f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157424
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29541}
In order to land the component build support in Chromium, it is
easier to turn on symbols export every time that is_component_build=true
instead of setting rtc_enable_symbol_export=is_component_build in
Chromium (since is_component_build is not available in .gn).
rtc_enable_symbol_export is still kept in the mix in order to turn
on symbol exports in any case a shared library will be added to the
WebRTC build.
Bug: webrtc:9419
Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29540}
All submodule pointers are now private.
The unique_ptr to a ApmPrivateSubmodules is replaced by a direct member
object.
The main outcome of this CL is that the code is nicer.
Bug: webrtc:5298
Change-Id: Ib8ef70a35a64b875752d2a318c572d152d51487a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157440
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29539}
merge two vectors of the same size into single vector
Remove redundant size_ variable.
Remove redundant variables in the StoredPacket internal struct.
Remove frame_created flags since shortly after it is set, used flag is set to false
Bug: webrtc:10979
Change-Id: Ia37944362abda4e2a6c6741f436f95c45e0f7069
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157174
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29535}
Local media SSRC is mandatory, but let's give it a default value to
make tests less brittle.
Bug: chromium:1015256
Change-Id: If7f6505482d90651bc58d9b358290c4d43487f4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157421
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29534}
Rename file with tests to match code under test.
Rename fixture by moving 'Test' from prefix to suffix
Bug: None
Change-Id: I54c36d3b517bde7cdffa3a7e74528cc464ea7ad7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157301
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29532}
This CL contains various cleanups/corrections to the multichannel AEC
code.
The changes have been shown to be bitexact over a large dataset.
Bug: webrtc:10913
Change-Id: Idd3e410b04527666e052f57ad81d0ac9eef3179b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157173
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29530}
This works on mobile and has less dependencies. There's no upside to
using gtest since I'm not planning on running the test anyway, so this
is a much better solution.
Bug: webrtc:11027
Change-Id: Id63af7086b9d9c9199c62bc8654b4202a4a1f759
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29529}
- RenderBuffer::Spectrum() loses its channel argument, allowing for
greater flexibility in passing the multi-channel spectrum data into
functions.
- The FFT spectra lengths are made compile-time constant, rendering
some DCHECKs obsolete.
Bug: webrtc:10913
Change-Id: Ied0c50cf72d974cfef7279fd2b9c572d049b8b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157104
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29528}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
