This change expands on https://webrtc-review.googlesource.com/c/src/+/374420 to cover the error produced when copying microphone samples.
Change-Id: I7aa58c9c9ac175d5f4cfdb60bbd3f16334c03c1b
Bug: webrtc:390314937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375540
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43826}
This adds an optional callback closure and an enum representing the error.
Bug: webrtc:390314937
Change-Id: If9a22dd6d90d5c4d94175e021511766ea49acec2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374420
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43780}
I already submitted separate CLs for /api and /components, here I'm
doing the rest of this folder
Format /sdk/objc/api folder
There are a lot of changes in /sdk so I'm splitting it
Formatting done via:
git ls-files | grep -E '^sdk\/objc\/.*\.(h|cc|mm)' | grep -Ev '^sdk\/objc\/(api|components).*' | xargs clang-format -i
No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: I8bb63f6696b2649ea642b900b9d9827980cc8315
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373884
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43696}
This change refactors existing self-assignments within if clauses across
the WebRTC codebase.
*Why:*
- Bug Prevention: Assignments within conditionals are frequently
unintended errors, often mistaken for equality checks.
- Clearer Code: Separating assignments from conditionals improves code
readability and reduces the risk of misinterpretation.
Change-Id: I199dc26a35ceca109a2ac569b446811314dfdf0b
Bug: chromium:361594695
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360460
Reviewed-by: Chuck Hays <haysc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42850}
There is a race condition on reading
`ObjCNetworkMonitor::_network_monitor` field.
The `ObjCNetworkMonitor::OnPathUpdate()` checks its nullability
on the org.webrtc.RTCDispatcherNetworkMonitor thread BEFORE the
`ObjCNetworkMonitor::Start()` assigns it on the network_monitor thread.
In addition, this field is neither atomic nor protected by mutex so its
last assigned value is not guaranteed to be visible to
another [reading] thread.
Bug: webrtc:355238623
Change-Id: I1a05215111cc873b7d4931824e18f281aebfb91f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357880
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42778}
This is a step towards sending audio timestamps from Meet in iOS.
Next step is to enable sending the audio timestamps (in harmony).
After enable absolute-capture-time header extension in harmony, the receiving participants will be able to store E2E audio latency and A/V sync metrics.
Bug: webrtc:13609
Change-Id: I797c1ed0035625ed065307314ac34c932c5abe7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334720
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41574}
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.
Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
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 9a0a6a198e8e247884fe01d7e0aa6bd425721c14
Original change's description:
> Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]"
>
> This is a reland of commit 2b9aaad58f56744f5c573c3b918fe072566598a5
>
> 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 9a0a6a198e8e247884fe01d7e0aa6bd425721c14.
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 2b9aaad58f56744f5c573c3b918fe072566598a5
>
> 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 2b9aaad58f56744f5c573c3b918fe072566598a5
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 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
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Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273340
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# 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}
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}
This reverts commit 8cd7b0a7babf3f58f6beab70fcabeb75e66c1bed.
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 02293096f9689fee3d32defa77dca227cc1eee90.
>
> 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 181ea6e414c5982015ce161e6368120be3658ec4.
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
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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}
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}
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}
