This is a space efficient way to store more records about decoded frames,
which is needed for long term references.
Bug: webrtc:9710
Change-Id: I051d59d34a966d48db011142466d9cd15304b7d9
Reviewed-on: https://webrtc-review.googlesource.com/c/116792
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26240}
Some callback wrappers set the callback_ member at the start, but
most set it after calling any owned implementation of Start().
Setting it after the call means that the callback_ is not set up
for any callbacks that happen during the call.
This cl fixes that by setting the callback_ member before any
calls are made in Start().
Bug: chromium:916961
Change-Id: Id26f8cc98377ef217f928095834162f5526c1fdf
Reviewed-on: https://webrtc-review.googlesource.com/c/117040
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
Cr-Commit-Position: refs/heads/master@{#26231}
This is a following up cl to the fix of crbug.com/911110. On Windows,
if an App window is suspended, it will block some queries (which
causes Chromium freezing and is fixed in Chromium.) and won't be captured.
So there is no reason to list it in the window capture picker.
Notes: this cl can't fix the case that the select app window becomes
non-responsive just before capturing starts. Hope that an extreme corner
case that can be safely ingored.
Bug: chromium:911110
Change-Id: I0d14872ac699d559f40b3bff70f048efc67ca5d9
Reviewed-on: https://webrtc-review.googlesource.com/c/115441
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26230}
A trimmed down version is moved to legacy_encoded_audio_frame_unittest.cc
where it's used for test parameterization.
Bug: webrtc:10185
Change-Id: I9abda22f9806b831b6ca4b27d6bcc888285f50f2
Reviewed-on: https://webrtc-review.googlesource.com/c/116961
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26218}
This reverts commit 6613f8e98ab3654ade7e8f5352d8d6711b157499.
Reason for revert: This change seemed innocent after all, so undoing speculative revert.
Original change's description:
> Revert "Refactor and remove media_optimization::MediaOptimization."
>
> This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
>
> Reason for revert: Speculative revert due to downstream crashes.
>
> Original change's description:
> > Refactor and remove media_optimization::MediaOptimization.
> >
> > This CL removes MediaOptmization and folds some of its functionality
> > into VideoStreamEncoder.
> >
> > The FPS tracking is now handled by a RateStatistics instance. Frame
> > dropping is still handled by FrameDropper. Both of these now live
> > directly in VideoStreamEncoder.
> > There is no intended change in behavior from this CL, but due to a new
> > way of measuring frame rate, some minor perf changes can be expected.
> >
> > A small change in behavior is that OnBitrateUpdated is now called
> > directly rather than on the next frame. Since both encoding frame and
> > setting rate allocations happen on the encoder worker thread, there's
> > really no reason to cache bitrates and wait until the next frame.
> > An edge case though is that if a new bitrate is set before the first
> > frame, we must remember that bitrate and then apply it after the video
> > bitrate allocator has been first created.
> >
> > In addition to existing unit tests, manual tests have been used to
> > confirm that frame dropping works as expected with misbehaving encoders.
> >
> > Bug: webrtc:10164
> > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26147}
>
> TBR=nisse@webrtc.org,sprang@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10164
> Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
> Reviewed-on: https://webrtc-review.googlesource.com/c/116780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26191}
TBR=nisse@webrtc.org,sprang@webrtc.org
Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/116960
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26213}
Comfort noise was generated by picking random angles on the unit circle
for each frequency band and then obtaining points on the unit circle from
{cos(a), -sin(a)}.
In order to reduce complexity, this change introduces a randomly indexed
table of 32 elements over sin(a). cos(a) is obtained by adding an offset
corresponding to pi/2 to the index. The table is pre-scaled by sqrt(2) to
avoid later multiplications.
This change reduces the computational complexity of AEC3 by ~8% with no
audible degradation.
Bug: webrtc:10189
Change-Id: I8cfe2469022fb1fe910ab3f966e55d9d499b7161
Reviewed-on: https://webrtc-review.googlesource.com/c/116787
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26209}
This will allow to increase the stored decoded frames history size and
optimize it to reduce memory consumption.
Bug: webrtc:9710
Change-Id: I82be0eb376c5d0b61ad5d754e6a37d606b4df29d
Reviewed-on: https://webrtc-review.googlesource.com/c/116686
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26200}
This is a preparation for deleting other modes than
VCMDecodeErrorMode::kNoErrors.
Bug: webrtc:8064
Change-Id: I614f8012f306c5d59e72bdb851b582c286cdd130
Reviewed-on: https://webrtc-review.googlesource.com/c/116781
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26195}
This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
Reason for revert: Speculative revert due to downstream crashes.
Original change's description:
> Refactor and remove media_optimization::MediaOptimization.
>
> This CL removes MediaOptmization and folds some of its functionality
> into VideoStreamEncoder.
>
> The FPS tracking is now handled by a RateStatistics instance. Frame
> dropping is still handled by FrameDropper. Both of these now live
> directly in VideoStreamEncoder.
> There is no intended change in behavior from this CL, but due to a new
> way of measuring frame rate, some minor perf changes can be expected.
>
> A small change in behavior is that OnBitrateUpdated is now called
> directly rather than on the next frame. Since both encoding frame and
> setting rate allocations happen on the encoder worker thread, there's
> really no reason to cache bitrates and wait until the next frame.
> An edge case though is that if a new bitrate is set before the first
> frame, we must remember that bitrate and then apply it after the video
> bitrate allocator has been first created.
>
> In addition to existing unit tests, manual tests have been used to
> confirm that frame dropping works as expected with misbehaving encoders.
>
> Bug: webrtc:10164
> Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26147}
TBR=nisse@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10164
Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
Reviewed-on: https://webrtc-review.googlesource.com/c/116780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26191}
This CL adds two unit tests to make sure that, when an echo path gain
change occurs, the echo canceller is notified.
Such a change can be caused by (i) a pre-amplifier gain change or
(ii) an analog gain change.
Bug: webrtc:7494
Change-Id: Ia47cfbbc5694340cd3e760d8d3c3393f79897a9d
Reviewed-on: https://webrtc-review.googlesource.com/c/111780
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26190}
When enabled, the delay manager is updated with reordered packets. It also makes the peak detector ignore the reordered packets.
Change-Id: I2bdc99764cc76b15e613ed3dc75f83aaf66eee4e
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/116481
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26187}
That means it does not have to be set on every update of StreamsConfig.
BUG=webrtc:9586
Change-Id: I6a348160e209042857c4475323466e2aa92adef8
Reviewed-on: https://webrtc-review.googlesource.com/c/116690
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26184}
This fixes a bug where we sometimes extract an Opus CNG packet and the packet after, even though there was big timestamp gap between the packets, which causes expansion during the next GetAudio calls.
Change-Id: I2409ac08df58afc496f74b91981657b7206e8bb1
Bug: webrtc:10167
Reviewed-on: https://webrtc-review.googlesource.com/c/115419
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26179}
That is, the payload packetization overhead (eg. vp8 payload header),
not the RTP headers, extensions, etc.
The encoder and pacer both look at payload rate, but are currently not
aware of the bytes that are added in between them.
Bug: webrtc:10155
Change-Id: I4cdb04849d762360374d47a496983c8c6df191d2
Reviewed-on: https://webrtc-review.googlesource.com/c/115410
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26163}
While this class is deprecated, it's needed as a stop-gap solution.
Other methods to configure the max probe rate all effect the current
estimate and/or trigger new probes to be sent, and we need a way to
configure the max without affecting other behavior.
Bug: webrtc:10070
Change-Id: I2b0ba2fef42d0bab6e5ea7f7c921681557802b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/114880
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26162}
This is a reland of 0cc42d47389c039c57e47d7ec0c76b97e2da2b0b
Original change's description:
> Reland "Default to dlopening the PipeWire."
>
> This is a reland of a099877d8946eb942046ca1295cc142e4fa7ea6f
>
> Original change's description:
> > Reland "Default to dlopening the PipeWire."
> >
> > This is a reland of a13be019017449c57f48203d0fb778f34f7553a7
> >
> > Original change's description:
> > > Default to dlopening the PipeWire.
> > >
> > > Reuse the existing infra from Chromium to do that. Additionally the
> > > target_gen_dir needs to the added to the include directories, otherwise
> > > the Chromium build will fail as it won't find the generated stubs. Also the
> > > pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> > > require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> > > doesn't work with them correctly. With all these changes in place the PipeWire
> > > support is enabled when compiling on Linux.
> > >
> > > Bug: chromium:682122
> > > Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > > Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> > > Cr-Commit-Position: refs/heads/master@{#25720}
> >
> > Bug: chromium:682122
> > Change-Id: I3cca3d4d961dc7a088346c8fd3c970d3dfde3b79
> > Reviewed-on: https://webrtc-review.googlesource.com/c/113040
> > Reviewed-by: Weiyong Yao <braveyao@chromium.org>
> > Reviewed-by: Brave Yao <braveyao@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25981}
>
> Bug: chromium:682122
> Change-Id: Ief26c93069f946f981340664a267fcb412229285
> Reviewed-on: https://webrtc-review.googlesource.com/c/114163
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26004}
Bug: chromium:682122
Change-Id: I0a4ea7b39be5970f26df6dbc3e437dd63cdb8708
Reviewed-on: https://webrtc-review.googlesource.com/c/116280
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26154}
This CL removes MediaOptmization and folds some of its functionality
into VideoStreamEncoder.
The FPS tracking is now handled by a RateStatistics instance. Frame
dropping is still handled by FrameDropper. Both of these now live
directly in VideoStreamEncoder.
There is no intended change in behavior from this CL, but due to a new
way of measuring frame rate, some minor perf changes can be expected.
A small change in behavior is that OnBitrateUpdated is now called
directly rather than on the next frame. Since both encoding frame and
setting rate allocations happen on the encoder worker thread, there's
really no reason to cache bitrates and wait until the next frame.
An edge case though is that if a new bitrate is set before the first
frame, we must remember that bitrate and then apply it after the video
bitrate allocator has been first created.
In addition to existing unit tests, manual tests have been used to
confirm that frame dropping works as expected with misbehaving encoders.
Bug: webrtc:10164
Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
Reviewed-on: https://webrtc-review.googlesource.com/c/115620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26147}
This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.
Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.
Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
When using WebRTC in iOS this Warning is shown for every single call even if stereo is not being used at all.
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Bug: webrtc:10146
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Reviewed-on: https://webrtc-review.googlesource.com/c/85283
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26133}
The ANA frame length controller requires the provided frame lengths supported by the encoder to be ordered. A data structural guarantee of such was in an earlier version but was accidentally removed since https://codereview.webrtc.org/2429503002. This CL uses std::set to ensure that again.
Change-Id: Ia197dbf6a34f02506e81c9f49d6cd60e4cdacef4
BUG: webrtc:6303
Reviewed-on: https://webrtc-review.googlesource.com/c/115946
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26119}
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct.
The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable().
Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible.
Bug: webrtc:9947
Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d
Reviewed-on: https://webrtc-review.googlesource.com/c/115243
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26101}
This reverts commit d1208c26b1cdb536fdec942207033711101d5d26.
Reason for revert: This cl causes the crashing issue as in
chromium:916961 at starting desktop capture on Windows.
Original change's description:
> Desktop capturer: Add OnDisplayChanged callback
>
> This adds support for a new DesktopCapturer::Callback method
> OnDisplayChanged that is sent at the start of a desktop capture
> session and whenever the display geometry changes.
>
> This cl adds the basic structure to call this api at the start
> of the capture session. Currently Windows only.
>
> A follow-up cl will add support to call this whenever the display
> geometry changes.
>
> Bug: webrtc:10122, chromium:915411
> Change-Id: Ie7283be5992454180daab1a60f58a3b2efdfed56
> Reviewed-on: https://webrtc-review.googlesource.com/c/114020
> Commit-Queue: Gary Kacmarcik <garykac@chromium.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26053}
TBR=jamiewalch@chromium.org,braveyao@webrtc.org,braveyao@chromium.org,garykac@chromium.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10122, chromium:915411, chromium:916961
Change-Id: Id0471e01bb90bb5accdf58262ae2b130cf343ecd
Reviewed-on: https://webrtc-review.googlesource.com/c/115433
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26095}
The new controller behaves mostly like before, but increases the target
rate on timer update rather than when feedback is received. This makes
the behavior easier to predict. It also uses a duration parameter to
track the increase, removing the meed for the minimum rate increase
constants that exists in the previous solution.
Bug: webrtc:9718
Change-Id: Iae31a9ba2d6474a8236f8eb72f86ff434f1d1fc6
Reviewed-on: https://webrtc-review.googlesource.com/c/114681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26088}
Just ignoring single_packet_reduction_len is wrong, because if the
fragment is put in a single packet it might still be the first or the
last packet in the whole sequence.
Bug: none
Change-Id: I4a2fbebe1d49cbef9298bb32d9cecaa617e4dfc3
Reviewed-on: https://webrtc-review.googlesource.com/c/115403
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26084}
