This is required by some virtual cameras, like Snap Camera from
Snapchat.
Bug: webrtc:14783
Change-Id: I3d841936c17f3f227af9a94a4c3b0f37940d43b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288361
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39073}
making it clear what unit is being used.
BUG=webrtc:13756
Change-Id: I6354d35a8e02bb93a905ccf32cb0b294b4813e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289460
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39008}
See commit
https://webrtc.googlesource.com/src/+/c8a6fb2bb8762de17008dee97c5fb6e762f7e056
where the setup methods for RTCCameraVideoCaptureTests' test cases were
lost. Both "setup" where XCTest instead looks for "setUp", and
"setupWithMockedCaptureSession" which isn't called explicitly anywhere.
This commit splits the old RTCCameraVideoCaptureTests into two;
RTCCameraVideoCaptureTests for tests using "setup", and
RTCCameraVideoCaptureTestsWithMockedCaptureSession for tests using
"setupWithMockedCaptureSession".
Bug: webrtc:8382
Change-Id: I64cefff744e12f62d65e04133512de1e10d17d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288601
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38931}
removes cached pixelbufferpool and instead retrieves current pool from the compressionSession each time (as recommended by apple docs)
Bug: webrtc:14688
Change-Id: I2244e69e7f32b912021db0905b9d5867d0bf6357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284240
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38920}
The file was renamed, see
https://groups.google.com/u/1/g/discuss-webrtc/c/ZQiP4f_bpw4
Bug: webrtc:14180
Change-Id: Ia76c85ba7d9da6b3a93d0a67a4b6a5187e07e230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283084
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38616}
It's deprecated and has been removed from Chrome. Let's follow suite.
// Passing all but unrelated bots
NOTRY=True
Bug: webrtc:14608
Change-Id: I6f2601af5b1dc08164230ebf15db2d2f1754f9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38488}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
Align thread checkers with the class comment,
i.e. ensure AudioDevice is used and destroyed on the same thread it was constructed on, not just the same thread AudioDevice::Init was called.
Bug: webrtc:9702
Change-Id: Ib905978cc8173266151adf26e1b7317f1d3852bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274164
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38018}
This is a reland of commit 9a0a6a198e
Original change's description:
> Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]"
>
> This is a reland of commit 2b9aaad58f
>
> 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: Iaf950d24bb2394a20e50421d5122f72ce46ae840
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273380
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37946}
Bug: webrtc:14193
Change-Id: I84a6462c233daae7f662224513809b13e7218029
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273662
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37977}
This reverts commit 9a0a6a198e.
Reason for revert: Breaks upstream project
Original change's description:
> Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]"
>
> This is a reland of commit 2b9aaad58f
>
> 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: Iaf950d24bb2394a20e50421d5122f72ce46ae840
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273380
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37946}
Bug: webrtc:14193
Change-Id: I5e18cc919ca4bb1cef7d5a11489451a0907f0d66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273486
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@google.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37950}
This is a reland of commit 2b9aaad58f
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: Iaf950d24bb2394a20e50421d5122f72ce46ae840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273380
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37946}
This reverts commit 2b9aaad58f.
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 reverts commit 83db78e854.
Reason for revert: Some tests in Chromium's blink no longer compile because of the change in the signature of the CreateDefaultTaskQueueFactory() function.
Original change's description:
> Add TaskQueueStdlib experiment.
>
> Bug: webrtc:14389
> Change-Id: I23c6e0ae675748ec35a99c334104dd2654995a33
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265802
> Commit-Queue: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37888}
Bug: webrtc:14389
Change-Id: If3e63d6b4ab9e838dc5020b88076a73fd29916e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272920
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Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37902}
The default implementation of CropAndScale uses ToI420() and then Scale,
and this implementation behaves inefficiently with RTCCVPixelBuffer.
Bug: None
Change-Id: I422ef80d124db0354a2e696892e882a78db445bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271140
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37877}
These tests were failing on mac-11 machines but seem to do fine on mac-12.
Bug: webrtc:13989,webrtc:13991
Change-Id: I11fb2302046fbb06b0824a4adc543a446405991b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272363
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#37843}
If the source image has a native handle and the encoder supports
the native handle, the encoder is expected to be able to correctly
sample/scale the source.
And VTCompressionSession can handle this, so DCHECK the frame
resolution only if the frame buffer is not native.
Bug: webrtc:14318
Change-Id: Id19c2f3bd86e9a2e1034d20e0255b4adc04a781f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270144
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37730}
Most calls to C++ PeerConnection and related classes are proxied
to internal threads in WebRTC. However, there is no such thing
in the Obj-C SDK.
It would be nice to proxy methods in the Obj-C SDK as well.
RTCMediaStream and RTCVideoTrack have NSMutableArray members,
and it can throw NSRangeException when it has race conditions,
so that it would be a good starting point.
Also, remove some NSAsserts as its condition isn't a fatal error,
and it doesn't affect the production already.
Bug: None
Change-Id: I10b44a9c773d62a5c04c254986733a6b67d51617
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37283}
This reverts commit 8cd7b0a7ba.
The assumption in AndroidNetworkMonitor that an interface name
is unique has turned out to be incorrect :( for some (weird) devices,
i.e ccmni0.
It is unclear if it is a permanent setup or a transient state.
This cl/ changes the impl. to cope with that, the last
OnNetworkConnected_n "owns" the interface name, and when
OnNetworkDisconnected_n runs, we check if we're "owner"
and maybe set a new "owner" (if we're not "owner" we do nothing).
New testcases added.
I also
1) change NetworkMonitorInterface to return a struct
with all the information that is requested with interface name
as key.
2) Change Network.cc adding (debug) assertions that network
properties can't change inside a loop (in one thread).
Original change's description:
> Revert "Reset all maps in AndroidNetworkMonitor Start()/Stop()"
>
> This reverts commit 02293096f9.
>
> Reason for revert: mysterious crashes in android_network_monitor.cc
>
> Original change's description:
> > Reset all maps in AndroidNetworkMonitor Start()/Stop()
> >
> > This cl/ fixes another race condition with the recent additions
> > to NetworkMonitorAutoDetect (getAllNetworksFromCache).
> >
> > The getAllNetworksFromCache-feature uses the by the Android team
> > preferred way of enumerating networks, i.e to register network listeners.
> >
> > Th recent fix to add IsAdapterAvailable, https://webrtc-review.googlesource.com/c/src/+/257400
> > contained a bug in that the adapter_type_by_name_ map was not
> > reset either on disconnect or Start/Stop.
> >
> > This cl/ addresses that including unit test.
> > It also de-obfuscates NetworkMonitor so that it always
> > calls NotifyOfActiveNetworkList on startMonitoring even
> > if list.size() == 0. This should not matter but makes
> > code easier to understand.
> >
> > Bug: webrtc:13741
> > Change-Id: I438b877eebf769a8b2e7292b697ef1c0a349b24f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258721
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36530}
>
> Bug: webrtc:13741
> Change-Id: I36fbf63f658d3e8048e13959cbebfbd14df12b14
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264146
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37016}
Bug: webrtc:13741
Change-Id: Ib4eb072b775e493b564528f0be94c685b70ec20f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264421
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37056}
This reverts commit 181ea6e414.
Reason for revert: Breaks downstream project. Kári will help to land it next week.
Original change's description:
> Add a prefix for objc category.
>
> According to the Google Objective-C style [1], category names should
> start with an appropriate prefix. WebRTC has some category definitions
> for system interfaces, but it doesn't use prefixes.
>
> $ otool -ov WebRTC.framework/WebRTC | grep -E "^[0-9a-z]{16} 0x[0-9a-z]+ __OBJC_._CATEGORY" | grep -v "_RTC"
> 0000000002160840 0x217c3c0 __OBJC_$_CATEGORY_UIDevice_$_H264Profile
> 0000000002160850 0x21808b8 __OBJC_$_CATEGORY_AVCaptureSession_$_DevicePosition
> 0000000002160858 0x2180968 __OBJC_$_CATEGORY_NSString_$_StdString
> 0000000002160860 0x21809c8 __OBJC_$_CATEGORY_NSString_$_AbslStringView
>
> To avoid conflicts, prefix the names and methods of those categories.
> Also remove sdk/objc/Framework/Classes/Common/NSString+StdString.h as
> it is not used by any other files.
>
> [1] https://google.github.io/styleguide/objcguide.html#category-naming
>
> Bug: webrtc:13884
> Change-Id: I2cf2742af198ab4e0bfb15c0476d72971e50ceee
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262341
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36880}
Bug: webrtc:13884
Change-Id: I85257088e4a3a62e01ff925ab5e77af83b078ef3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262420
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
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Cr-Commit-Position: refs/heads/main@{#36885}
According to the Google Objective-C style [1], category names should
start with an appropriate prefix. WebRTC has some category definitions
for system interfaces, but it doesn't use prefixes.
$ otool -ov WebRTC.framework/WebRTC | grep -E "^[0-9a-z]{16} 0x[0-9a-z]+ __OBJC_._CATEGORY" | grep -v "_RTC"
0000000002160840 0x217c3c0 __OBJC_$_CATEGORY_UIDevice_$_H264Profile
0000000002160850 0x21808b8 __OBJC_$_CATEGORY_AVCaptureSession_$_DevicePosition
0000000002160858 0x2180968 __OBJC_$_CATEGORY_NSString_$_StdString
0000000002160860 0x21809c8 __OBJC_$_CATEGORY_NSString_$_AbslStringView
To avoid conflicts, prefix the names and methods of those categories.
Also remove sdk/objc/Framework/Classes/Common/NSString+StdString.h as
it is not used by any other files.
[1] https://google.github.io/styleguide/objcguide.html#category-naming
Bug: webrtc:13884
Change-Id: I2cf2742af198ab4e0bfb15c0476d72971e50ceee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262341
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36880}
The tests in rtc_unittests_objc are old gtest based tests and
The tests in sdk_unittests are XCTest based tests.
The objc tests in rtc_unittest are causing problems [1],
so I think it's time to combine the two types of objc tests.
Renaming the files to match the existing sdk_unittests and
removing the use of gtest helper functions (eg, EXPECT_EQ)
are planned for follow-up CLs.
[1] https://webrtc-review.googlesource.com/c/src/+/261724/9
Bug: webrtc:8382
Change-Id: I9308b35f654c4479c9df72bf7cd7f4032e267eb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36878}
In https://webrtc-review.googlesource.com/c/src/+/244420,
I added sanity check of RTCVideoFrame in RTCMTLVideoView, but I forgot
to modify the related tests.
Fix this by adding the appropriate property stubs to RTCVideoFrame stubs
created in RTCMTLVideoViewTests.
Bug: webrtc:13990, webrtc:13490
Change-Id: I21f0f75cd052e4255e1eee5f39ffdd50c2f8e61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260420
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36714}
Fixing a race condition where session.sampleRate changes before AudioDeviceIOS::HandleValidRouteChange() finishes.
session.sampleRate is read into session_sample_rate at 576 and used at 623 to initialize the audio unit. However, in the call to SetupAudioBuffersForActiveAudioSession() the session.sampleRate is read again and may have changed, resulting in different sample rates used for the buffers and the audio unit. The consequence is a sample rate mismatch with either high pitched or low pitched audio.
The fix is to always use the buffer sample rate for the audio unit.
The DCHECK at 622 would save us in debug, but not in production, hence removed.
Change-Id: I562f1bf7f94d7447139ada2636b02ade7fcd6371
Bug: webrtc:14011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260329
Reviewed-by: Henrik Andreasson <henrika@google.com>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36708}
The goal of this CL is to make the CFDictionaryRef not dependent
on a fixed number of properties, which will facilitate future work.
No behavior change intended.
Bug: None
Change-Id: I32261d81eaa9b77380cecbdaefcbaeafde300f9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257920
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36449}
convert rtc_base/network and collateral.
This also remove last usage of system_wrappers/field_trials
in p2p/...Yay!
Bug: webrtc:10335
Change-Id: Ie8507b1f52bf7f3067e9b4bf8c81a825e4644fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36357}
This is part of a large-scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL converts the majority of "const std::string&"s in function
parameters under rtc_base/ to absl::string_view.
Bug: webrtc:13579
Change-Id: I2b1e3776aa42326aa405f76bb324a2d233b21dca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254081
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Anders Lilienthal <andersc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36239}
This is part of a large scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL adds absl::string_view versions of the OnLogMessage functions in
rtc::LogSink. The const std::string& versions are kept for now since
downstream clients use subclasses of LogSink and need to be migrated
before these are removed.
Bug: webrtc:13579
Change-Id: I57bb72ad503805ff0ca16f1d7aece2d44c65cb73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Owners-Override: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36189}
This is a reland of 325789c457
Original change's description:
> Mark all bool conversion operators as explicit
>
> An explicit bool conversion operator will still be used implicitly
> when an expression appears in "bool context", e.g., as the condition
> in an if statement, or as argument to logical operators. The
> `explicit` annotation prevents conversion in other contexts, e.g.,
> converting both a and b to bool in an expression like `a == b`.
>
> Bug: None
> Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35927}
Bug: None
Change-Id: Ie057dfc8c0b5c498e2c8daff7620172c89f0e011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35962}
This reverts commit 325789c457.
Reason for revert: Breaks downstream clients.
Original change's description:
> Mark all bool conversion operators as explicit
>
> An explicit bool conversion operator will still be used implicitly
> when an expression appears in "bool context", e.g., as the condition
> in an if statement, or as argument to logical operators. The
> `explicit` annotation prevents conversion in other contexts, e.g.,
> converting both a and b to bool in an expression like `a == b`.
>
> Bug: None
> Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35927}
TBR=mbonadei@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I392cd0c7bd96c90e0db20831864418adb7d58bc3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251080
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
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Cr-Commit-Position: refs/heads/main@{#35929}
An explicit bool conversion operator will still be used implicitly
when an expression appears in "bool context", e.g., as the condition
in an if statement, or as argument to logical operators. The
`explicit` annotation prevents conversion in other contexts, e.g.,
converting both a and b to bool in an expression like `a == b`.
Bug: None
Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35927}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
This CL also removed the temporary enum value NotSpecified.
See PSA https://groups.google.com/u/1/g/discuss-webrtc/c/SdoVP02eUIk
for more information.
Bug: webrtc:11121
Change-Id: Ib19e1f5911ffad001fc61ac28174eb8e823fc803
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246208
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35759}
Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.
They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.
Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.
Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
Updates all webrtc code, to have a small followup cl to just add the
"explicit" keyword. Patchset #24 passed all webrtc tests, with explicit.
Bug: webrtc:13464
Change-Id: I39863d3752f73209b531120f66916dc9177bf63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242363
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35718}
the new spelling is more standard and more compact, in particular doesn't need extra include and thus dependency
Bug: None
Change-Id: Iaea69d2154e4d9eff2468514f5734cb3fe016ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245080
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35709}
Added Nutanix Inc. to the AUTHORS file.
PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
Bug: chromium:1251096
Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35684}
The default value of sdpSemantics is about to change from PlanB to
UnifiedPlan. In order not to cause subtle bugs by applications that
depend on the default value being PlanB, we are temporarily making the
default NotSpecified. Constructing with NotSpecified causes the C++
layer to crash (https://webrtc-review.googlesource.com/c/src/+/242968).
This is in accordance to the publically announced plans:
https://groups.google.com/u/1/g/discuss-webrtc/c/SdoVP02eUIk
Bug: webrtc:11121
Change-Id: Idbb8fd0f5c224311cf1f25ac2832800124ed14d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246060
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35678}
In preparation for switching the default from kPlanB to kUnifiedPlan,
which could cause subtle bugs for those not prepared for it, we change
the default to kNotSpecified. The only purpose of kNotSpecified is to
crash, forcing any dependencies to explicitly set their sdp_semantics
value.
Tests are updated to explicitly set sdp_semantics when necessary, and
where the test does not care we update to kUnifiedPlan.
If this change lands without getting reverted we can let it sit for a
few weeks, after which we should change the default to kUnifiedPlan and
delete kNotSpecified.
Bug: webrtc:11121
Change-Id: I19b669b0735d78e269e19eaae86c2d7d95a91141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242968
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35651}
This also removes all internal usage of RemoveTrack, and changes
the replacement function to RemoveTrackOrError rather than RemoveTrackNew.
Bug: webrtc:9534
Change-Id: Idf7bb17495686de77c70428dcbfb12278328ce59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244094
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35624}
This CL adds the callback on ICE Candidate Error to the Android and
the iOS SDKs.
Spec: https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-onicecandidateerror
Bug: webrtc:13446
Change-Id: I6e511aaa80f1aa8f4310d8518d1144d97470cd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239460
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35531}
In logging.cc, use the pointer of the static variable so that
it doesn't need a global constructor/exit time destructor.
In RTCFieldTrials.mm, store the field trial string as a char pointer
instead of a std::unique_ptr to ensure that it is never freed.
LSAN will be unhappy with this fix, but WebRTC itself hasn't been
tested with LSAN enabled, and any code changed in this CL does not
build with build_with_chromium=true, so it shouldn't be a problem.
Bug: webrtc:9693, webrtc:11665
Change-Id: Ia28e3534170e0817b815717f6efe862f7b51ef62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237320
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35391}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Error message "Tried to get DTMF sender from video sender." should no
longer pollute logs.
Bug: None
Change-Id: I60d6f45ba049e93ec06d645da43fb8269354edf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235982
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#35322}
setExif: would create a CFDictionary using NULL for keyCallBacks and
valueCallBacks. This has the effect of comparing the keys of the
dictionary by pointer instead of by value. With ld64, this works
because it always dedupes identical constant CFSTR("foo") literal,
but lld currently doesn't do this.
Using kCFTypeDictionaryKeyCallBacks and kCFTypeDictionaryValueCallBacks
fixes the problem with lld and is "more correct" in general: Now the
dictionary would work with computed CFStrings too, it shows up better
in CFShow() output, etc.
While here, also fix a memory leak in setExif:.
Bug: chromium:1251763
Change-Id: I43c96d2189a4a77fe3bd0dfb3e33623925b0f900
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35067}
This CL is for the same behavior as before [1], to emit the NSError
when an application is not using the RTCAudioSession lock correctly.
[1] https://webrtc-review.googlesource.com/c/src/+/207432
Bug: webrtc:13091
Change-Id: I031b0e963d33c92ce1af7a306edfa6be005e043d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229461
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35018}
With this change, we catch audio unit start errors and pipe them to the
audio session. The audio session notifies its delegate, which can then
take appropriate action based on the error code.
The signal follows the same path as the playout glitch detection.
Bug: webrtc:13119
Change-Id: I8c9f9d2a1e3457447d0ce61ad197f7e1c6392837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230240
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34862}
