Accessing this method from the test thread is illegal,
but doesn't always fail.
Bug: webrtc:8524
Change-Id: Ie0e84cc2fb63268fb6d7cbf0c3a58cb35312c16b
Reviewed-on: https://webrtc-review.googlesource.com/49061
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21930}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
Mark functions with override instead of virtual.
Add explicit non-trivial constructors/assign operators/destructors.
Define them in .cc files instead of inlining
use auto* instead of auto when deduced type is raw pointer
Bug: webrtc:163
Change-Id: I4d8a05d6a64fcc2ca16d02c5fcf9488fda832a6d
Reviewed-on: https://webrtc-review.googlesource.com/48781
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21927}
This is a reland of 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=sprang@webrtc.org
Bug: webrtc:8653
Change-Id: I32fa92649f3ff40b1e364f880040e52ae698f74d
Reviewed-on: https://webrtc-review.googlesource.com/46860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21918}
This CL adds robustness in terms of echo removal and faster recovery
in order to regain echo canceller transparency after echo path changes.
The CL does:
-Improve the adaptation rate of the linear filter.
-Increase the look-window used before the linear filter has adapted.
-Decrease the effects of missed detection of residual echo.
-Increase the safety margin before allowing the suppressor gain to
increase.
Bug: chromium:804873,webrtc:8788
Change-Id: I28eedc4c8d0a4f0bc7b79c02d6d59bf00fddd566
Reviewed-on: https://webrtc-review.googlesource.com/48721
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21917}
Rtp Packets in webrtc expected to be less that 1500,
i.e. way less that 2^16 bytes for extensions block.
This CL explicitly discards longer extension.
Bug: chromium:809046
Change-Id: Ibed33b51bafc3fd4804ec135f66110c6d2796734
Reviewed-on: https://webrtc-review.googlesource.com/48061
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21910}
This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
Bug: webrtc:8415
Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da
Reviewed-on: https://webrtc-review.googlesource.com/48000
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21899}
WHAT: made a BUILD.gn with library and tests in the Audio Processing
Module Voice Activity Detector directory. Updated depending
code. Fixed a Clang warning.
WHY: to make it possible for a target to depend on just the VAD and
not the whole APM. There are other benefits:
* Sometimes faster compilation.
* The VAD takes up 28000 bytes of libjingle_peerconnection_so.so. Making
a peerconnection shared object file without the VAD has to be done in
steps. The first step is a custom target for the VAD. Hence this Cl.
Change-Id: Iea0207a0b5979db26baaf46b24beaefbb1c431af
BUG: webrtc:5716, webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/47521
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21893}
This helps separate concerns, so that the VideoProcessorIntegrationTest
is almost oblivious to the fact that it needs to connect to the JVM
to get the Android HW codecs.
Bug: webrtc:8448
Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8
Reviewed-on: https://webrtc-review.googlesource.com/47384
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21890}
This reverts commit 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
Reason for revert: Major regressions on perf bots.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ia8a273eb9e92b7d0d960c49658c228208170962d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/47560
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21877}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I8a7ab64dfecdb3da4099fdec61e5fc27af4f8ccc
Reviewed-on: https://webrtc-review.googlesource.com/47380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21874}
So we can report perf results using JSON and not parsing stdout.
I reordered the way the arguments are parsed, so that options go
at the end, and not at the middle, which is an awkward place to put them.
Regular usage specifying [-I], bottleneck_value, infile and outfile
shouldn't be affected.
Bug: chromium:807737
Change-Id: Ida863846400326c33e443d723f384971b891b6e5
Reviewed-on: https://webrtc-review.googlesource.com/47161
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21873}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I36f01784fa5f5b77eefc02db479b1f7f6ee1a8c3
Reviewed-on: https://webrtc-review.googlesource.com/46263
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21871}
The goal of this work is to make it easier to experiment with the
bandwidth estimation implementation. For this reason network control
functionality is moved from SendSideCongestionController(SSCC),
PacedSender and BitrateController to the newly created
GoogCcNetworkController which implements the newly created
NetworkControllerInterface. This allows the implementation to be
replaced at runtime in the future.
This is the first part of a split of a larger CL, see:
https://webrtc-review.googlesource.com/c/src/+/39788/8
For further explanations.
Bug: webrtc:8415
Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
Reviewed-on: https://webrtc-review.googlesource.com/43840
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21868}
The VideoCodecTest class is a fixture base class for the
libvpx-VP8, libvpx-VP9, and OpenH264 unit tests. It is unrelated
to the VideoProcessor tests, which we colloquially refer to as
the "codec test".
This rename is thus to reduce this confusion. It should have no
functional impact.
Bug: webrtc:8448
Change-Id: If73443bda5df0f29a71ce6ce069ac128795ff0ad
Reviewed-on: https://webrtc-review.googlesource.com/47160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21867}
Merge OveruseDetector into the TrendlineEstimator (send side BWE only) and remove the OveruseDetector from DelayBasedBwe. The behavior should be the same as before. One expection is that if no packets were received for kStreamTimeOutMs (2 seconds), it would previously reset the trendline estimator but not the detector. Since they have been merged, it now resets both.
Create an interface that the estimators will implement to facilitate experimentation with different estimators/detectors.
Bug: webrtc:8729
Change-Id: I5c3d2161a0d0dcb2e8a140c0fd887f0286d70fd4
Reviewed-on: https://webrtc-review.googlesource.com/38781
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21852}
This can be used to wrap Objective-C components in C++ classes, so users
can use the WebRTC C++ API directly together with the iOS specific
components provided by our SDK.
Bug: webrtc:8832
Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8
Reviewed-on: https://webrtc-review.googlesource.com/46162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21850}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I1fc4cb50d81522a486888a626d4a95df7847d591
Reviewed-on: https://webrtc-review.googlesource.com/46743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21849}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I6c345c38fd990f66bc1a8129e7f7cee7d161e926
Reviewed-on: https://webrtc-review.googlesource.com/47120
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21848}
WebRTC internal code should always used include paths that starts
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc
Reviewed-on: https://webrtc-review.googlesource.com/46101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21847}
Encoded frames are preserved and decoded after all layers are
encoded.
Each spatial layer is decoded with separate decoder.
For quality evaluation of lowres layers original input frame is
downscaled with bilinear interpolation.
Encoded and decoded frames are dumped into separate files.
For async codecs encoded frames are passed to decoder in encode
callback, as before.
Bug: webrtc:8524
Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1
Reviewed-on: https://webrtc-review.googlesource.com/43381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21844}
This reverts commit 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.
Reason for revert: breaks downstream projects
Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}
TBR=deadbeef@webrtc.org,sprang@webrtc.org,shampson@webrtc.org
Change-Id: If14074a7fc56c83b75584d8e9a6a913a40514bad
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8653
Reviewed-on: https://webrtc-review.googlesource.com/46840
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21832}
Due to libvpx we were restricted to always turning the low simulcast
stream on, or else the encoder would always label the active streams'
encoded frames as key frames. Now that libvpx has been updated and
rolled in, this change updates tests to reflect that it is working.
Bug: webrtc:8653
Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
Reviewed-on: https://webrtc-review.googlesource.com/46340
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21831}
Due to the growing number of arguments, these functions are being replaced by the AudioProcessingBuilder class.
Bug: webrtc:8668
Change-Id: Ic3936fbd47d92eac22a857a678dca5fd8c029d8b
Reviewed-on: https://webrtc-review.googlesource.com/46241
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21826}
This fix the mismatch of resolution VP9 wrapper set for coded frame with
its actual resolution.
Bug: webm:1485, webrtc:5749
Change-Id: Ie1225d8f3a3d00e66229a1a79858d0a89b3d5fae
Reviewed-on: https://webrtc-review.googlesource.com/46040
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21819}
This is a temporary measure until the synchronization method
used in the class, gets fixed.
Bug: chromium:796889, chromium:795340
Change-Id: Ie3d394ae42f005e8e0f353d04ea9c1d053ea9fd2
Reviewed-on: https://webrtc-review.googlesource.com/40460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21812}
The original rtc_event_log_api is refactored to a pure API target plus
multiple targets coupled with WebRTC implementations.
Bug: None
Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
Reviewed-on: https://webrtc-review.googlesource.com/43247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#21811}
