This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae
... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909
... and 2072b87261a6505a88561bdeab3e7405d7038eaa
Reason for revert: Failing DuoGroupsMediaQualityTest due to missing
TaskQueuePacedSender::EnsureStarted() in google3.
Fix: This CL adds the logic behind TaskQueuePacedSender::EnsureStarted,
but initializes with |is_started| = true. Once the caller in google3 is
updated, |is_started| can be switched to false by default.
> Original change's description:
> Reason for revert: crashes due to uninitialized pacing_bitrate_
> crbug.com/1190547
> Apparently pacer() is sometimes being used before EnsureStarted()
> Fix: Instead of delaying first call to SetPacingRates(),
> this CL no-ops MaybeProcessPackets() until EnsureStarted()
> is called for the first time.
> Original change's description:
> > [Battery]: Delay start of TaskQueuePacedSender.
> >
> > To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> > only upon RtpTransportControllerSend::EnsureStarted().
> >
> > More specifically, the repeating task happens in
> > TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> > task_queue_.PostDelayedTask().
> >
> > Bug: chromium:1152887
> > Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> > Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33421}
>
> Bug: chromium:1152887
> Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#33554}
Bug: chromium:1152887
Change-Id: Ie365562bd83aefdb2757a65e20a4cf3eece678b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213000
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33629}
libaom is compiled with REALTIME_ONLY option. Soon it will be impossible
to create encoder or request default config with usage other than
AOM_USAGE_REALTIME. Fixing the wrapper to use proper usage parameter
Bug: None
Change-Id: I862741a724e4a8524f22ae79700b3da6517dbfb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214100
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33624}
This change adds basic support for setting codecType kVideoCodecAV1 in
VCMEncodedFrames.
Bug: chromium:1191972
Change-Id: I258b39ff89c8b92ebbb288ef32c88b900a35d10e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213182
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33594}
This reverts commit c184047fef005b86a6dd76f03b0eb5ec01de3c5c.
Reason for revert: Breaks the WebRTC->Chromium roll:
ERROR Unresolved dependencies.
//third_party/webrtc/test/fuzzers:vp9_encoder_references_fuzzer(//build/toolchain/win:win_clang_x64)
needs //third_party/webrtc/modules/video_coding:mock_libvpx_interface(//build/toolchain/win:win_clang_x64)
We need to add tryjob to catch these. The fix is to make
//third_party/webrtc/modules/video_coding:mock_libvpx_interface
visible in built_with_chromium builds by moving the target
out of this "if" https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/video_coding/BUILD.gn;l=615;drc=3889de1c4c7ae56ec742fb9ee0ad89657f638169.
Original change's description:
> Add fuzzer to validate libvpx vp9 encoder wrapper
>
> Fix simulcast svc controller to reuse dropped frame configuration,
> same as full svc and k-svc controllers do.
> This fuzzer reminded the issue was still there.
>
> Bug: webrtc:11999
> Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33568}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I1676986308c6d37ff168467ff2099155e8895452
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212973
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33573}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
Bug: webrtc:11999
Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33568}
This reverts commit 2072b87261a6505a88561bdeab3e7405d7038eaa.
Reason for revert: Causing test failure.
Original change's description:
> Reland "[Battery]: Delay start of TaskQueuePacedSender." Take 2
>
> This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae
> ... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909
>
> Reason for revert: crashes due to uninitialized pacing_bitrate_
> crbug.com/1190547
> Apparently pacer() is sometimes being used before EnsureStarted()
> Fix: Instead of delaying first call to SetPacingRates(),
> this CL no-ops MaybeProcessPackets() until EnsureStarted()
> is called for the first time.
>
> Original change's description:
> > [Battery]: Delay start of TaskQueuePacedSender.
> >
> > To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> > only upon RtpTransportControllerSend::EnsureStarted().
> >
> > More specifically, the repeating task happens in
> > TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> > task_queue_.PostDelayedTask().
> >
> > Bug: chromium:1152887
> > Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> > Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33421}
>
> Bug: chromium:1152887
> Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#33554}
TBR=hbos@webrtc.org,sprang@webrtc.org,etiennep@chromium.org
Change-Id: I430fd31c7602702c8ec44b9e38e68266abba8854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1152887
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212965
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33559}
This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae
... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909
Reason for revert: crashes due to uninitialized pacing_bitrate_
crbug.com/1190547
Apparently pacer() is sometimes being used before EnsureStarted()
Fix: Instead of delaying first call to SetPacingRates(),
this CL no-ops MaybeProcessPackets() until EnsureStarted()
is called for the first time.
Original change's description:
> [Battery]: Delay start of TaskQueuePacedSender.
>
> To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> only upon RtpTransportControllerSend::EnsureStarted().
>
> More specifically, the repeating task happens in
> TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> task_queue_.PostDelayedTask().
>
> Bug: chromium:1152887
> Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33421}
Bug: chromium:1152887
Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33554}
Follow-up CL to VP8 and VP9 encoders taking care of mapping.
Context again:
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
In this CL, VideoStreamEncoder no longer calls GetMappedFrameBuffer() on
behalf of the encoders, since the encoders are now able to either do the
mapping or performs ToI420() anyway.
- Tests for old VSE behaviors are updated to test the new behavior (i.e.
that native frames are pretty much always forwarded).
- The "having to call ToI420() twice" workaround to Android bug
https://crbug.com/webrtc/12602 is added to H264 and AV1 encoders.
Bug: webrtc:12469
Change-Id: Ibdc2e138d4782a140f433c8330950e61b9829f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211940
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@{#33548}
`RTCInboundRtpStreamStats.lastPacketReceivedTimestamp` must be a time
value in milliseconds with Unix epoch as time origin (see
bugs.webrtc.org/12605#c4).
This change fixes both audio and video `RTCInboundRtpStreamStats` stats.
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12605
Change-Id: I68157fcf01a5933f3d4e5d3918b4a9d3fbd64f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33547}
This will further speed up intra frame encoding
Bug: None
Change-Id: I3c836502cdcb1037e3128850a085b92acd8fc7ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212821
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Cr-Commit-Position: refs/heads/master@{#33544}
This patch adds support for sending zero video layer allocations
header extensions. This can be used to signal that a stream is
turned off.
Bug: webrtc:12000
Change-Id: Id18fbbff2216ca23179c58ef7bbe2ebea5e242af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212743
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33541}
This is a follow-up to the VP9, fixing VP8 this time. Context again:
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
- To achieve this, WebRTC encoders are updated to map kNative video
buffers so that in a follow-up CL VideoStreamEncoder can stop mapping
intermediate buffer sizes.
Bug: webrtc:12469, chromium:1157072
Change-Id: I026527ae77e36f66d02e149ad6fe304f6a8ccb05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212600
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@{#33537}
Namespace used because of copy-pasting an old pattern, should never have been used in the first place. Removing it now to make followup refactoring prettier.
Bug: webrtc:12579
Change-Id: I00a80958401cfa368769dc0a1d8bbdd76aaa4ef5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212603
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33536}
WgcCaptureSession would crash when copying the frame data for an image
from a portrait oriented monitor. This is because we were using the
height of the image multiplied by the rowpitch of the buffer to
determine the size of the data to be copied. However, in portrait
mode the height measures the same dimension as the rowpitch, leading
to us overrunning the frame buffer.
The fix is to use the height and width of the image multiplied by
the number of bytes per pixel to determine how much data to copy
out of the buffer, and only use the rowpitch to advance the pointer
in the source data buffer. This has the added benefit of giving us
contiguous data, reducing the size of the DesktopFrame that we output.
Bug: webrtc:12490
Change-Id: I4c26f8864cb57ac566a742af70fea1da504b9706
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209501
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33532}
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
- To achieve this, WebRTC encoders are updated to map kNative video
buffers so that in a follow-up CL VideoStreamEncoder can stop mapping
intermediate buffer sizes.
In this CL LibvpxVp9Encoder is updated to map kNative buffers of pixel
formats it supports and convert ToI420() if the kNative buffer is
something else. A fake native buffer that keeps track of which
resolutions were mapped, MappableNativeBuffer, is added.
Because VP9 is currently an SVC encoder and not a simulcast encoder, it
does not need to invoke CropAndScale.
This CL also fixes MultiplexEncoderAdapter, but because it simply
forwards frames it only cares about the pixel format when
|supports_augmented_data_| is true so this is the only time we map it.
Because this encoder is not used with kNative in practise, we don't care
to make this path optimal.
Bug: webrtc:12469, chromium:1157072
Change-Id: I74edf85b18eccd0d250776bbade7a6444478efce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212580
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@{#33526}
This will further speed up intra frame encoding
Bug: None
Change-Id: I1a105c6d2cdd9dc82f84d0039dbea3f0d090ab93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212320
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Cr-Commit-Position: refs/heads/master@{#33492}
This is a reland of aa6adffba325f4b698a1e94aeab020bfdc47adec
What was changed in the reland is that the merging of the bands is
excluded from the code that is not run when the output is not used.
I.e., the merging is always done.
This is important to have since some clients may apply muting before APM,
and still flag to APM that the signal is muted. If the merging is not
always done, those clients will get nonzero output from APM during muting.
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: Ib74dd1cefa173d45101e26c4f2b931860abc6d08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33478}
This CL adds functionality in the noise suppressor that allows the
computational complexity to be reduced when the output of APM is not used.
Bug: b/177830919
Change-Id: I849351ba9559fae770e4667d78e38abde5230eed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211342
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33477}
This CL adds functionality in AEC3 that allows the computational
complexity to be reduced when the output of APM is not used.
Bug: b/177830919
Change-Id: I08121364bf966f34311f54ffa5affbfd8b4db1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211341
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33476}
This will speed up key frame encoding (together with libaom changes)
3x-4x times with ~13% BDRate loss on key frames only
Bug: None
Change-Id: I24332f4f7285811cdc6619ba29844fe564cae95e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212040
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Commit-Queue: Fyodor Kyslov <kyslov@google.com>
Cr-Commit-Position: refs/heads/master@{#33468}
This CL adds functionality that allows adjusting the audio levels
internally in APM. The main purpose of the functionality is to allow
APM to optionally be moved to an integration that does not provide an
analog gain to control, and the implementation of this has been
tailored specifically to meet the requirements for that.
More specifically, this CL does
-Add a new variant of the pre-amplifier gain that is intended to replace
the pre-amplifier gain (but at the moment can coexist with that). The
main differences with the pre-amplifier gain is that an attenuating
gain is allowed, the gain is applied jointly with any emulated analog
gain, and that its packaging fits better with the post gain.
-Add an emulation of an analog microphone gain. The emulation is
designed to match the analog mic gain functionality in Chrome OS (which
is digital) but should be usable also on other platforms.
-Add a post-gain which is applied after all processing has been applied.
The purpose of this gain is for it to work well with the integration
in ChromeOS, and be used to compensate for the offset that there is
applied on some USB audio devices.
Bug: b/177830918
Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33466}
This CL adds one frame (10 ms) of silence in APM output after unmuting to mask
audio resulting from the turning on the processing that was deactivated
during the muting.
Bug: b/177830919
Change-Id: If44cfb0ef270dde839dcd3f0b98d1c91e81668dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33454}
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats`
via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams.
`GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and
`ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`.
Bug: webrtc:12529
Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33452}
A refactoring (https://webrtc-review.googlesource.com/c/src/+/196520)
of decoder metadata handling introduced a bug which causes us to log an
info-level entry for every frame decoded if the implementation changes
during runtime (e.g. due to software fallback).
This CL fixes that to avoid spamming the logs.
Bug: webrtc:12271
Change-Id: I89016351b8752b259299c4cf56c6feddcca43460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211664
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33451}
This reverts commit aa6adffba325f4b698a1e94aeab020bfdc47adec.
Reason for revert: breaks webrtc-importer
Original change's description:
> Reduce complexity in the APM pipeline when the output is not used
>
> This CL selectively turns off parts of the audio processing when
> the output of APM is not used. The parts turned off are such that
> don't need to continuously need to be trained, but rather can be
> temporarily deactivated.
>
> The purpose of this CL is to allow CPU to be reduced when the
> client is muted.
>
> The CL will be follow by additional CLs, adding similar functionality
> in the echo canceller and the noiser suppressor
>
> Bug: b/177830919
> Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33431}
Bug: b/177830919
Change-Id: I937cd61dedcd43150933eb1b9d65aebe68401e91
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211348
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33433}
This CL selectively turns off parts of the audio processing when
the output of APM is not used. The parts turned off are such that
don't need to continuously need to be trained, but rather can be
temporarily deactivated.
The purpose of this CL is to allow CPU to be reduced when the
client is muted.
The CL will be follow by additional CLs, adding similar functionality
in the echo canceller and the noiser suppressor
Bug: b/177830919
Change-Id: I72d24505197a53872562c0955f3e7b670c43df6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209703
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33431}
