Reverted the logic introduced in https://codereview.webrtc.org/2933953003
This is because the audio device buffer behavior changed with https://codereview.webrtc.org/3009193002, and the RequestPlayoutData method now returns number of samples in each channel, which creates mismatch the reverted CL.
Bug: webrtc:8548
Change-Id: Id4711ca48437ddd3483327c2a4c7827d09e5b770
Reviewed-on: https://webrtc-review.googlesource.com/24122
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20737}
This CL refactors the delay estimator in AEC3.
Furthermore, it adds:
1. Allow for a customized delay estimator behavior to simplify
development.
2. Exposes that behavior to clear configuration settings.
3. Adds logging of the delay range supported by the delay
estimator.
Bug: webrtc:8519
Change-Id: I1764a090519a78b021b2e7de565c52a6c02c848e
Reviewed-on: https://webrtc-review.googlesource.com/21166
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20733}
Because some hardware encoders may not preserve capture timestamp, use
rtp timestamp as a key to capture time storage and restore corrupted
capture timestamps whenever possible.
Bug: webrtc:8497
Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4
Reviewed-on: https://webrtc-review.googlesource.com/23620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20732}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=kwiberg@webrtc.org
Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20728}
This makes it easier to import cleanly downstream, and makes it
a lot easier to reason about.
Bug: webrtc:6828
Change-Id: I953e129de73053f8619333fe7e318b36e3a1fffa
Reviewed-on: https://webrtc-review.googlesource.com/22722
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20727}
int32_t SetLoudspeakerStatus(bool enable)
int32_t GetLoudspeakerStatus(bool* enabled) const
These APIs are only implemented on iOS and they do not belong in the
native audio layer since the client can achieve the same functionality
by using the shared audio session in sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h.
It also gives the client a better flexibility in how the audio routing is done.
Bug: webrtc:7306
Change-Id: I853e2f57e0f5ae0a0f9fc4729ce961d81f92588b
Reviewed-on: https://webrtc-review.googlesource.com/23740
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20721}
The RtcpBandwidthObserverImpl did not provide any features that a raw pointer does not have. deprecating it to simplify further refactoring down the line. Preferring raw pointer usage as it provides equivalent functionality in less code.
Bug: webrtc:8415
Change-Id: Id2c4c73a331835f124da8d308615ca2ce34b2d1b
Reviewed-on: https://webrtc-review.googlesource.com/22500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20712}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=terelius@webrtc.org
Bug: None
Change-Id: I1d2538482e2390d91f3285124d011a578da4b61b
Reviewed-on: https://webrtc-review.googlesource.com/23564
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20710}
Use moving median filters in RemoteNtpTimeEstimator, RtpToNtpEstimator
If Webrtc-ClockEstimation experiment is enabled, median filtering is
applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
out random errors introduced by incorrect RTCP SR reports and networking delays.
Bug: webrtc:8468
Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595
NOTRY=TRUE
NOPRESUBMIT=TRUE
Change-Id: I592c4083fefc0dbdebe7b3ff30b92e95ba337595
Reviewed-on: https://webrtc-review.googlesource.com/23263
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20708}
Ran into these when trying a newer libstdc++
Bug: None
Change-Id: Ie3ce0ae1ae1e6da1a15476fbf942b48b37adc9fa
Reviewed-on: https://webrtc-review.googlesource.com/23501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20701}
If Webrtc-ClockEstimation experiment is enabled, median filtering is
applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
out random errors introduced by incorrect RTCP SR reports and networking
delays.
Bug: webrtc:8468
Change-Id: Iec6d094d2809d1efeb0b9483059167d9a03880da
Reviewed-on: https://webrtc-review.googlesource.com/22682
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20682}
VideoEncoderSoftwareFallbackWrapper is updated to take a VideoEncoder as
argument instead relying on built-in SW codecs. The purpose is to make
VideoEncoderSoftwareFallbackWrapper more modular and not depend on
built-in SW encoders.
Bug: webrtc:7925
Change-Id: I99896f0751cfb77e01efd29c97d3bd07bdb2c7c0
Reviewed-on: https://webrtc-review.googlesource.com/22320
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20671}
In preparation for a new remote clock estimation:
RtpToNtpEstimator will now store unwrapped rtp timestamps in
|measurements| and |calculated| flag is out of |Parameters| struct.
Additional missing |Reset| method added to MovingMedianFilter and
PercentileFilter
Bug: webrtc:8468
Change-Id: I1bb454ba0c93720cbb30bfded19677daaf9f721c
Reviewed-on: https://webrtc-review.googlesource.com/22020
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20669}
Conditional visibility is complex to maintain and it is not well
supported by other build systems.
This CL removes it and falls back on the more relaxed visibility value
("*" in this case).
It is not a problem because the targets that are using conditional
visibility are all marked as "testonly" and this is probably enough to
keep the build graph clean.
Bug: None
Change-Id: I2d2b5ac9a02d08c2863950116db455976ee1459c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14902
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20658}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
- Add alpha accessors to PlanarYuvBuffer interface, null by defualt.
- Add WrapI420ABuffer() that creates a container which implements these
accessors.
- Show the use via StereoDecoderAdapter.
This CL is the step 2 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Id5691cde00088ec811b63d89080d33ad2d6e3939
Reviewed-on: https://webrtc-review.googlesource.com/21130
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20635}
This excludes these frames from being counted as dropped by encoder.
Also fix bitrate projected distribution for vp9 svc for outliers
detection.
Bug: webrtc:8497
Change-Id: Id37487456170c61e2323a660668f0c319ea5831d
Reviewed-on: https://webrtc-review.googlesource.com/21223
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20632}
Even if we're not going to transmit any timing info over the wire.
Bug: webrtc:8504
Change-Id: Id54192a10e6b2a6a2cb57a2ff6b28fc0d16e471d
Reviewed-on: https://webrtc-review.googlesource.com/21160
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20628}
This reverts commit 267d84baf0597f89a3d1f66d323db754bc5d9239.
Reason for reland: Fix the bug; decoder is not allowed to ever be null and we need to use a
NullVideoDecoder that ignores calls instead.
Original change's description:
> Revert "Update internal video decoder factory to new interface"
>
> This reverts commit b2fc9b1b104240e68047901309deaee3e8b94bea.
>
> Reason for revert: Suspected to cause failures on Android bots on webrtc.fyi, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/21051
>
> Original change's description:
> > Update internal video decoder factory to new interface
> >
> > We want to move away from cricket::WebRtcVideoDecoderFactory and this CL
> > updates the internal factory. Also, VideoDecoderSoftwareFallbackWrapper
> > is updated to take a VideoDecoder as argument instead of a factory so it
> > can be used with external SW decoders.
> >
> > Bug: webrtc:7925
> > Change-Id: Ie6dc6c24f8610a2129620c6e2b42e3cebb2ddef7
> > Reviewed-on: https://webrtc-review.googlesource.com/7301
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20597}
>
> TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org
>
> Change-Id: I0a12c98fdc30f00d58c85ee7e088f50160d39724
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/21420
> Reviewed-by: Christian Fremerey <chfremer@webrtc.org>
> Commit-Queue: Christian Fremerey <chfremer@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20605}
TBR=brandtr@webrtc.org,magjed@webrtc.org,andersc@webrtc.org,chfremer@webrtc.org,chfremer@google.com
Change-Id: I6cf5794dc3fadfa86809a94da80b69dbb4c56f52
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/21541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20623}
