Replace most instances. SetAlrStartTime is set as is should be cleaned up together with the callsite.
Bug: webrtc:14404
Change-Id: I8ec532828ef665afbf08f0943465a429ab40baa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37932}
This reverts commit 2b9aaad58f56744f5c573c3b918fe072566598a5.
Reason for revert: Breaks upstream project
Original change's description:
> ObjC ADM: record/play implementation via RTCAudioDevice [3/3]
>
> # Overview
> This CL chain exposes new API from ObjC WebRTC SDK to inject custom
> means to play and record audio. The goal of CLs is achieved by having
> additional implementation of `webrtc::AudioDeviceModule`
> called `ObjCAudioDeviceModule`. The feature
> of `ObjCAudioDeviceModule` is that it does not directly use any
> of OS-provided audio APIs like AudioUnit, AVAudioEngine, AudioQueue,
> AVCaptureSession etc. Instead it delegates communication with specific
> system audio API to user-injectable audio device instance which
> implements `RTCAudioDevice` protocol.
> `RTCAudioDevice` is new API added to ObC WebRTC SDK in the CL chain.
>
> # AudioDeviceBuffer
> `ObjCAudioDeviceModule` does conform to heavy `AudioDeviceModule`
> interface providing stubs for unrelated methods. It also implements
> common low-level management of audio device buffer, which glues audio
> PCM flow to/from WebRTC.
> `ObjCAudioDeviceModule` owns single `webrtc::AudioDeviceBuffer` which
> with the help of two `FineAudioBuffer` (one for recording and one for
> playout) is exchanged audio PCMs with user-provided `RTCAudioDevice`
> instance.
> `webrtc::AudioDeviceBuffer` is configured to work with specific audio:
> it has to know sample rate and channels count of audio being played and
> recorded. These formats could be different between playout and
> recording. `ObjCAudioDeviceModule` stores current audio parameters
> applied to `webrtc::AudioDeviceBuffer` as fields of
> type `webrtc::AudioParameters`. `RTCAudioDevice` has it's own variable
> audio parameters like sample rate, channels count and IO buffer
> duration. The audio parameters of `RTCAudioDevice` must be kept in sync
> with audio parameters applied to `webrtc::AudioDeviceBuffer`, otherwise
> audio playout and recording will be corrupted: audio is sent only
> partially over the wire and/or audio is played with artifacts.
> `ObjCAudioDeviceModule` reads current `RTCAudioDevice` audio parameters
> when playout or recording is initialized. Whenever `RTCAudioDevice`
> audio parameters parameters are changed, there must be a notification to
> `ObjCAudioDeviceModule` to allow it to reconfigure
> it's `webrtc::AudioDeviceBuffer`. The notification is performed
> via `RTCAudioDeviceDelegate` object, which is provided
> by `ObjCAudioDeviceModule` during initialization of `RTCAudioDevice`.
>
> # Threading
> `ObjCAudioDeviceModule` is stick to same thread between initialization
> and termination. The only exception is two IO functions invoked by SDK
> user code presumably from real-time audio IO thread.
> Implementation of `RTCAudioDevice` may rely on the fact that all the
> methods of `RTCAudioDevice` are called on the same thread between
> initialization and termination. `ObjCAudioDeviceModule` is also expect
> that the implementation of `RTCAudioDevice` will call methods related
> to notification of audio parameters changes and audio interruption are
> invoked on `ObjCAudioDeviceModule` thread. To facilitate this
> requirement `RTCAudioDeviceDelegate` provides two functions to execute
> sync and async block on `ObjCAudioDeviceModule` thread.
> Async block could be useful when handling audio session notifications to
> dispatch whole block re-configuring audio objects used
> by `RTCAudioDevice` implementation.
> Sync block could be used to make sure changes to audio parameters
> of ADB owned by `ObjCAudioDeviceModule` are notified, before interrupted
> playout/recording restarted.
>
> Bug: webrtc:14193
> Change-Id: I5587ec6bbee3cf02bad70dd59b822feb0ada7f86
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269006
> Reviewed-by: Henrik Andreasson <henrika@google.com>
> Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
> Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37928}
Bug: webrtc:14193
Change-Id: I6e759a91664c1f6f60e862d72e45f75c51d7297a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273340
Auto-Submit: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Owners-Override: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37931}
# Overview
This CL chain exposes new API from ObjC WebRTC SDK to inject custom
means to play and record audio. The goal of CLs is achieved by having
additional implementation of `webrtc::AudioDeviceModule`
called `ObjCAudioDeviceModule`. The feature
of `ObjCAudioDeviceModule` is that it does not directly use any
of OS-provided audio APIs like AudioUnit, AVAudioEngine, AudioQueue,
AVCaptureSession etc. Instead it delegates communication with specific
system audio API to user-injectable audio device instance which
implements `RTCAudioDevice` protocol.
`RTCAudioDevice` is new API added to ObC WebRTC SDK in the CL chain.
# AudioDeviceBuffer
`ObjCAudioDeviceModule` does conform to heavy `AudioDeviceModule`
interface providing stubs for unrelated methods. It also implements
common low-level management of audio device buffer, which glues audio
PCM flow to/from WebRTC.
`ObjCAudioDeviceModule` owns single `webrtc::AudioDeviceBuffer` which
with the help of two `FineAudioBuffer` (one for recording and one for
playout) is exchanged audio PCMs with user-provided `RTCAudioDevice`
instance.
`webrtc::AudioDeviceBuffer` is configured to work with specific audio:
it has to know sample rate and channels count of audio being played and
recorded. These formats could be different between playout and
recording. `ObjCAudioDeviceModule` stores current audio parameters
applied to `webrtc::AudioDeviceBuffer` as fields of
type `webrtc::AudioParameters`. `RTCAudioDevice` has it's own variable
audio parameters like sample rate, channels count and IO buffer
duration. The audio parameters of `RTCAudioDevice` must be kept in sync
with audio parameters applied to `webrtc::AudioDeviceBuffer`, otherwise
audio playout and recording will be corrupted: audio is sent only
partially over the wire and/or audio is played with artifacts.
`ObjCAudioDeviceModule` reads current `RTCAudioDevice` audio parameters
when playout or recording is initialized. Whenever `RTCAudioDevice`
audio parameters parameters are changed, there must be a notification to
`ObjCAudioDeviceModule` to allow it to reconfigure
it's `webrtc::AudioDeviceBuffer`. The notification is performed
via `RTCAudioDeviceDelegate` object, which is provided
by `ObjCAudioDeviceModule` during initialization of `RTCAudioDevice`.
# Threading
`ObjCAudioDeviceModule` is stick to same thread between initialization
and termination. The only exception is two IO functions invoked by SDK
user code presumably from real-time audio IO thread.
Implementation of `RTCAudioDevice` may rely on the fact that all the
methods of `RTCAudioDevice` are called on the same thread between
initialization and termination. `ObjCAudioDeviceModule` is also expect
that the implementation of `RTCAudioDevice` will call methods related
to notification of audio parameters changes and audio interruption are
invoked on `ObjCAudioDeviceModule` thread. To facilitate this
requirement `RTCAudioDeviceDelegate` provides two functions to execute
sync and async block on `ObjCAudioDeviceModule` thread.
Async block could be useful when handling audio session notifications to
dispatch whole block re-configuring audio objects used
by `RTCAudioDevice` implementation.
Sync block could be used to make sure changes to audio parameters
of ADB owned by `ObjCAudioDeviceModule` are notified, before interrupted
playout/recording restarted.
Bug: webrtc:14193
Change-Id: I5587ec6bbee3cf02bad70dd59b822feb0ada7f86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269006
Reviewed-by: Henrik Andreasson <henrika@google.com>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37928}
This adds a (default off) flag which makes retransmissions be processed
immediately, just like audio packets normally are.
This might increase send rates and thus losses in some cases, but will
also reduce retranmission delays especially when timer slack or bursting
is used. Usefuleness TBD via experiment.
Bug: chromium:1354491
Change-Id: Icaa83125bfb30826ce72e6e786963d411e05ea57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272483
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37926}
- If there are more than 5 IPv6 networks, then diversify IPv6 interface types selection.
- Passing field_trial from peer_connection_factory.cc when creating BasicPortAllocator object.
Bug: webrtc:14334
Change-Id: I7d100d944f4e60414e3421f422997bc3f168cc24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271581
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37924}
Add field trial to not probe if loss based limited
If both Alr probing and periodic probing of networkstate estimate is enabled, probes are limited by the network state estimate * factor controlled by field trial.
Bug: webrtc:14392
Change-Id: I46e1dbdd8b14f63a7c223b4c03c114717b802023
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272805
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37915}
- this builder is excluded from tree closers.
Since this is the very first builder that uses reclient, it may fail for misconfiguration.
Feel free to revert if it blocks critical tasks.
Bug: b:243628179
Change-Id: Ia7535f181d3cb195cd030670899e1730f06500c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272869
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Takuto Ikuta <tikuta@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Junji Watanabe <jwata@google.com>
Cr-Commit-Position: refs/heads/main@{#37909}
This tests all existing codecs (VP8, VP9) with the depdendency
descriptor and adds the AV1 tests that requires it as well.
Placeholders for missing modes have been added for both VP9 and AV1.
Bug: webrtc:11607
Change-Id: Ie900bddc54ccbf4dcc466f3a7a6c8241906a243a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272807
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37906}
This simplifies the logic of what simulcast layers to signal, and avoids
situations where the upper layers get confused about which layers exist.
Bug: chromium:1350245
Change-Id: I9edeb93cbb30e872c4d3f3429a85a1fccf17996a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37905}
And add a unit test that verifies that next probe time is set at is
the expected if the recommended probe size is used.
Bug: webrtc:14392
Change-Id: I239bb3a1c8eefc85509aacc82037c64e3ce49ed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272648
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37901}
The libyuv fix for this issue was submitted > 2 months ago so it
should be safe to remove the workaround from webrtc.
Bug: chromium:1330019
Change-Id: Ibcf3818739673005e40d7ef9917c5f5692c50df4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272861
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37895}
rtc::Thread::Dispose is only used in test code,
but complicates the main thread loop.
Bug: webrtc:8324
Change-Id: I2dccdadcdc932b9992958d1e70fb93d1879b7618
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272821
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37894}
I would like to run a separate capturer for each desktop on Linux and
I ran into the DCHECK in XErrorTrap when I was prototyping that
solution. I addressed it by using a Mutex and then experienced and
occasional hang when capturing which I traced down to
SharedXDisplay::ProcessPendingXEvents(), this is a shared display
instance used by each unique capturer instance so I added a mutex
there as well.
I ran 2 capturer instances concurrently for well over an hour and did
not experience any hangs or capture artifacts.
Bug: webrtc:2022
Change-Id: Ia6778cae4bbae48886fe45f2991f02e0ea08fef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271920
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/main@{#37892}
Remote Peek function as unused
Move Get and Dispatch into private section to ensure they are not used
from outside.
Bug: webrtc:9702
Change-Id: Ibd0b236fe43543d60f97f988524526493bbeaaa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272804
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37889}
Fix reporting of time_between_freezes_ms when all peers were
unregistered before end of the call.
Bug: b/243501613
Change-Id: Ie2b77d8399e4f4bf672e80f62ed27609336af171
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272650
Reviewed-by: Andrey Logvin <landrey@google.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37885}
