The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.
Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
Before this, an empty list of CSRCs was always provided up to encoded
insertable streams transforms for remote video tracks, regardless of
the actual CSRCs on received frames. Audio already works correctly.
Bug: chromium:1411614
Change-Id: I51ab4dc5e67a1a35893fefff16c1f057e9047e6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291539
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39240}
The mouse_cursor_monitor_unittest.cc was disabled on all platforms, so
it can be deleted.
Bug: webrtc:3408
Change-Id: I294bc502993a5b0a369a60a751c72f72ec909dfc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291724
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39239}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
This is a prerequisite step to break apart AudioCodingModule and AcmReceiver.
Bug: webrtc:14867
Change-Id: Iba589c7a31b6346ff4acb727793d84077162c8c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291534
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39235}
For frames captured and sent to the callback immediately, we
are not sending the capturer ID as we used to do in base capturer
pipewire. Adding the capturer id as well as the frame capture time
so as to keep the sent frame to be in sync with the
non-immediate-frame-send implementation.
Bug: chromium:1291247
Change-Id: I02693907928b9e770ea56f89b46c37f17f4bc4a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291680
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39228}
Today, behaviour is decided based on if transport sequence number v2 is
in the SDP answer. But it might be better to decide based on received
packets since it is valid to negotiate both extensions.
Another bonus With this solution is that Call does not need to know
about receive header exensions.
This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337
Bug: webrtc:7135
Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39226}
Earlier, we were relying on CRD to periodically do frame captures.
This is however not needed since Wayland captures are event based
and once the compositor has send the event/frame to us, we can just
handover the frame to a registered callback. This ensures that we
have a single frame clock that (i.e. one present in the compositor).
Without this change, there is a chance that CRD capture clock is run
out of sync with the compositor's clock and cause unnecessary frame
delays.
See the following ideal scenario, for example, where '+' indicates
when a frame is sent by the compositor and when CRD tries to capture
it. This assumes that the same clock cycle for both CRD and the
compositor, each cycle length is enclosed within | .... |).
Compositor Frame Ready | +... | | +... |
CRD Frame Capture | .+.. | | .+.. |
In this case, when both the clocks are in sync, CRD is able to
capture frame right after it is generated by the compositor. But if
they are completely out of sync, then CRD can always see a stale
frame (delayed by one cycle and it can cause users to feel stutter).
Compositor Frame Ready | .+.. | | .+.. |
CRD Frame Capture | +... | | +... |
This stutter can become worse if the compositor is delayed in
generating the frames for some reason (e.g. load on the system).
Bug: chromium:1291247
Change-Id: I7c10c724fbbd87dc523d474e7ece8e8aa146fd7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291080
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39218}
This change adds support for renegotiating the frame rate
via pipewire.
Bug: chromium:1291247
Change-Id: Iacd4a3c924900839a8db75a50b448df6c48c83ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291460
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39216}
Allow CSRCs to be modified per-frame in an Encoded Insertable Streams
transform, to support a web API which allows per-frame CSRC
modifications to signal when a JS application has changed the source
of the video which is written into an encoded frame.
Initially only for Video, with Audio support likely to follow later.
Bug: webrtc:14709
Change-Id: Ib34f35faa9cee56216b30eaae42d7e65c78bb9f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291324
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tove Petersson <tovep@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39214}
This is to avoid using 0 as a default value.
Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp.
Bug: webrtc:13609
Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118
Reviewed-by: Olov Brändström <brandstrom@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39177}
- Lowering the energy threshold for updating the accumulated error.
- Not using the pre-echo estimate in the initial frames when the matched filters have been recently initialized.
- Slight speed up for the increases in the accumulated error.
- Not periodically resetting the accumulated error.
Bug: webrtc:14205
Change-Id: Ic337332e263b27d7a3aba0ab4b371517780f9c90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291320
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39175}
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}
