Based on offline testing; needed to allow input volume adaptations
more frequently. Note that if the estimated speech level falls in
the target range, the recommended input volume won't change and
hence the new lower threshold won't necessarily increase the
number of adjustments.
Bug: webrtc:7494
Change-Id: Iabb501c188da238ea7b7137175bcfe09239c90a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291110
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39161}
This is used by CRD and export is required for component builds
to work properly.
Bug: chromium:1291247
Change-Id: I281e490b7d00cbd074b96eac905976af38400f8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291200
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39143}
This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate.
Bug: webrtc:14818
Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39123}
When the `WebRTC-Audio-GainController2` field trial is used, the
initial APM configuration is adjusted depending on its original
values and the field trial parameters.
This CL fixes two cases when the code crashes:
1. when, in the initial APM config, AGC1 is enabled, AGC2 is
disabled and TS is enabled
2. when the initial APM sample rate is different from the
capture one and the VAD APM sub-module is not re-initialized
This CL also improves the unit tests coverage and it has been
tested offline to check that the VAD sub-module is created only
when expected and that AGC2 uses its internal VAD when expected.
The tests ran on a few Wav files with different sample rates and
one AEC dump and on 16 different APM and field trial
configurations.
Bug: chromium:1407341, b/265112132
Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39118}
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
RTPVideoHeader is changed to non-const to allow modifying it. We want
to do this when implementing setMetadata() in JavaScript or when
refactoring clone() as "construct + set bytes + setMetadata".
Unblocks
https://chromium-review.googlesource.com/c/chromium/src/+/4164979.
Bug: webrtc:14709
Change-Id: I6089df9c03e9aa33feeb0830dd240dd456cb565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39113}
This is now ready for plumbing to Chromium layers.
Once it's exposed in JavaScript (behind flag!) we can evaluate whether
all of this information is really needed or if the information is
superflous (e.g. already contained in the raw bytes).
Bug: webrtc:14709
Change-Id: I3837ef86046704a300ec8a108c8c9477bd91b9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290884
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39102}
This will allow exposing VP8, VP9 and H264-specific RTP header metadata
in JavaScript (behind a flag).
This information appears to be necessary for cloning
(https://github.com/w3c/webrtc-encoded-transform/issues/161), and
cloning should be the same as "new frame + setMetadata + setBytes",
ergo this should be exposed.
Bug: webrtc:14709
Change-Id: Ie71c05f40689bbd529dc4674a07a87c7910b22d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39101}
This is needed for Chromium. The video capture API in Chromium expects the
raw frames and it will always convert or copy the frame. With the existing
API that would mean copying the frame twice.
Bug: webrtc:13177
Change-Id: I71f6e2dc6d5a812c3641ac691b75d50178fa0de7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264548
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39095}
PacketArrivalMap explicitly doesn't promise packet at the beginning
of it is received. Ensuring that property is wasteful
Bug: chromium:1382563
Change-Id: Ifc898b7ec2bc7a302af8dcfd233e0c598f62db95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290501
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39083}
Fallback to a default value if the scalability mode is unset or not supported by the codec.
The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set).
Bug: webrtc:11607
Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39074}
We found that the legacy assumption for H264 which assumed that
simulcast streams would use 2x width ratios in unnecessary as the
encoder has since been fixed to handle multiple ratios.
H264 encoder still works even if this assumption is invalid
Bug: None
Change-Id: I9caacf78d26c8215b94858a2d8674ec4cd64e96e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286940
Reviewed-by: Mirta Dvornicic <mirtad@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39072}
to remove duplicated implementation of these functions between av1 packetizer, av1 depacketizer and video allocation rtp header extension
Bug: None
Change-Id: I30049f31c289bdb9e0aad6520f5145d1f999e635
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290731
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39069}
RtpPacket has CopyOnWriteBuffer and std::vector that can be moved more
efficiently than copied, thus move of the RtpPacket is also more efficient
Bug: None
Change-Id: I5509346e426cd32d0fb0649ef1a6883b7176df1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290726
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39053}
After having generated one second of comfort noise and not received any packets, switch to expand mode which will fade out to silence and enter the efficient muted mode.
The behavior is enabled by default but can be disabled through a field trial.
Bug: webrtc:12790
Change-Id: I1e2c1acced3e4a2c1c1595824f1303a0c339aeb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290578
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39043}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
rtc::CopyOnWriteBuffer::SetSize extends buffer with uninitialized memory by design.
It is up to the user of the rtc::CopyOnWriteBuffer to ensure it is initialized.
Bug: chromium:1404299
Change-Id: I41f3f91bf20ff440984d78ed81e01f5db36ff509
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38972}
rtc::CopyOnWriteBuffer::SetSize extends buffer with uninitialized memory by design.
It is up to the user of the rtc::CopyOnWriteBuffer to ensure it is initialized.
Bug: chromium:1403397
Change-Id: Ic0111a84bda32379770ddb1c7d24bee10d96b7a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289041
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38959}
which needs to be taken into account separately if the
primary SSRC has been acknowledged but the RTX SSRC has
not.
If nothing has been acknowledged, mid+rid are sent on the primary SSRC and mid+rrid are sent on the RTX SSRC.
If the primary SSRC has been acknowledged, no extensions are sent on the primary SSRC and mid+rrid are sent on the RTX SSRC.
If both the primary SSRC and the RTX SSRC have been ack'd, no extensions are sent on either primary or RTX SSRC.
BUG=webrtc:13896
Change-Id: Ice251fae23a881ee9c9edc71b5d5c45a32ac76d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38949}
Instead of getting header extension mapping from a receiver object, get the mapping from the received packet.
The purpose is to be able to remove extension information from webrtc/call/receive_stream.h.
Header extensions are negotiated per mid, not per receive stream.
The goal is to reduce the number of places where packets are parsed and demuxed.
Bug: webrtc:7135, webrtc:14795
Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38944}
This CL records the time it took to capture a frame.
Bug: chromium:1291247
Change-Id: I31cbb2ca6ae5b9449b8fd154182105a3ce2c851e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288660
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38933}
- Test behavior with no input volume controller
- Test behavior with startup volume higher than the minimum
input volume
Bug: webrtc:7494
Change-Id: I36d48e2bd277b8a71eb6fbb0272c26c7176b3d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286380
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38932}
Make sure that the input volume controller implementations exhibit
the adaptive behavior regardless of the sample rate and the number
of channels. The newly added tests check that:
- a downward adjustment takes place with clipping input
- an upward adjustment takes place with a too low speech level
- a downward adjustment takes place with a too high speech level
Bug: webrtc:14761
Change-Id: I1795e74c5f219e15107e928ebaca2bfa75214526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287301
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38930}
