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
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
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}
This CL removes the last "nogncheck" comment that was related to a
known build cycle. The remaining ones are because of conditional
dependencies.
Bug: webrtc:8733
Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37302}
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 6e4d7e606c.
Reason for revert: Still breaks downstream build (though in a different way this time)
Original change's description:
> Reland "Delete old Android ADM."
>
> This is a reland of commit 4ec3e9c988
>
> Original change's description:
> > Delete old Android ADM.
> >
> > The schedule move Android ADM code to sdk directory have been around
> > for several years, but the old code still not delete.
> >
> > Bug: webrtc:7452
> > Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37174}
>
> Bug: webrtc:7452
> Change-Id: Icabad23e72c8258a854b7809a93811161517266c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37236}
Bug: webrtc:7452
Change-Id: Ide8fbd55fadd7aed9989053afff7c63c04f1320f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266023
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37242}
This is a reland of commit 4ec3e9c988
Original change's description:
> Delete old Android ADM.
>
> The schedule move Android ADM code to sdk directory have been around
> for several years, but the old code still not delete.
>
> Bug: webrtc:7452
> Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37174}
Bug: webrtc:7452
Change-Id: Icabad23e72c8258a854b7809a93811161517266c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265872
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37236}
This reverts commit d609473b9c.
Reason for revert: Breaks downstream project
Original change's description:
> Move Java PeerConnectionFactory.fieldTrialsFindFullName to different file.
>
> Currently, Java equivalent of webrtc::field_trial::FindFullName is in
> PeeerConnectionFactory, which belongs to peerconnection_java GN target.
>
> Move that method into a separate file and GN target to make it easier
> to use the fieldTrialsFindFullName method in other code.
>
> Bug: webrtc:13973
> Change-Id: I4d7d30339883af76c1d066f72270c6caf9c64c49
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261500
> Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
> Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37233}
Bug: webrtc:13973
Change-Id: I08eda44444aee4d64a0cee36f3f921f75088d7fc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265922
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37234}
Currently, Java equivalent of webrtc::field_trial::FindFullName is in
PeeerConnectionFactory, which belongs to peerconnection_java GN target.
Move that method into a separate file and GN target to make it easier
to use the fieldTrialsFindFullName method in other code.
Bug: webrtc:13973
Change-Id: I4d7d30339883af76c1d066f72270c6caf9c64c49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261500
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37233}
This reverts commit 4ec3e9c988.
Reason for revert: Causes downstream build error.
Original change's description:
> Delete old Android ADM.
>
> The schedule move Android ADM code to sdk directory have been around
> for several years, but the old code still not delete.
>
> Bug: webrtc:7452
> Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37174}
Bug: webrtc:7452
Change-Id: If094e0a3ef5a3d340cbd5dfa0a8a88c3e97ba0bf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265393
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37180}
The schedule move Android ADM code to sdk directory have been around
for several years, but the old code still not delete.
Bug: webrtc:7452
Change-Id: I0f75c680f71f0b2ce614de6cbd9f124c2a59d453
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264620
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37174}
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 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}
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>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
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}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
When an SCTP stream is closing, a stream reset needs
to be sent from both ends.
The remote was not sending a stream reset and quickly
opening another stream with the same StreamID could
cause SCTP errors.
Bug: webrtc:13994
Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36773}
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}
This better reflects the ownership passing of AddTrack, and is more
consistent for RemoveTrack.
Bug: webrtc:13980
Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36603}
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
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}
Modify cl/ a bit and add fieldtrialsstring on observer
not to break downstream projects.
---
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 14/inf
This cl/ passes field trials all the way from c++
to the android NetworkMonitorAutoDetect.java
Bug: webrtc:10335
Change-Id: Ic6842612eed36b684340f0f78f4087bee249cc50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36498}
---
Bug: webrtc:10335
Change-Id: Ied43770977465a0042541a61d29a9015c0b9cdc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258622
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36520}
A managed device might have camera access restricted, which results in
the following crash:
Caused by android.os.ServiceSpecificException: validateClientPermissionsLocked:1044: Callers from device user 0 are not currently allowed to connect to camera "1"
at android.os.Parcel.createException(Parcel.java:2085)
at android.os.Parcel.readException(Parcel.java:2039)
at android.os.Parcel.readException(Parcel.java:1987)
at android.hardware.ICameraService$Stub$Proxy.connectDevice(ICameraService.java:624)
at android.hardware.camera2.CameraManager.openCameraDeviceUserAsync(CameraManager.java:389)
at android.hardware.camera2.CameraManager.openCameraForUid(CameraManager.java:588)
at android.hardware.camera2.CameraManager.openCamera(CameraManager.java:516)
at org.webrtc.Camera2Session.openCamera(Camera2Session.java:359)
at org.webrtc.Camera2Session.start(Camera2Session.java:322)
at org.webrtc.Camera2Session.<init>(Camera2Session.java:298)
at org.webrtc.Camera2Session.create(Camera2Session.java:276)
at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:214)
at android.os.HandlerThread.run(HandlerThread.java:67)
Change-Id: I5e7b8d238e9381d1f8a4fe9858e8eb480d578fa0
Bug: webrtc:13950
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258363
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36513}
This cl/ passes field trials all the way from c++
to the android NetworkMonitorAutoDetect.java
Bug: webrtc:10335
Change-Id: Ic6842612eed36b684340f0f78f4087bee249cc50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36498}
The two GN templates are nearly identical, so merge them to reduce
maintenance.
Bug: webrtc:13949
Change-Id: I5f53ade5f9d09ce6f23a6cb29c9d39df4485a237
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36495}
Move global state into a singleton class. Eliminates use of
GlobalMutex. Also use std::atomic rather than volatile, for improved
thread safety.
Bug: webrtc:11567
Change-Id: I305d16ed2f4a9a6497df119e66d9bba08337339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258120
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36475}
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}
This CL replaces those references with the smallest set of targets
that can satisfy the linker dependencies revealed by building the
"all" target.
Bug: webrtc:13634
Change-Id: Ia778630b18e1164138c41d245c3c8effed67f8e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257282
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36445}
This cl/ fixes a 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.
This however introduces a unpleasant race condition like this:
1) network.cc discover rmnet0
2) BasicPortAllocator tries to create UDP port on rmnet0
- This fails as BindSocketToNetwork requires a android handle.
3) NetworkMonitorAutoDetect gets callback with rmnet0
4) BasicPortAllocator tries to create TCP port on rmnet0
- This succeeds.
5) Since rmnet0 has one working port, there will not be
any new ports created on that network
=> We end up with a TCP only connection :(
---
By impl. IsAdapterAvailable, we make sure that the network
will not be used by BasicPortAllocator (or anyone else!)
until we support binding to it.
The IsAdapterAvailable was implemented for IOS,
and has test coverage using FakeNetworkManager.
This cl/ is default enabled with the kill-switch
WebRTC-AndroidNetworkMonitor-IsAdapterAvailable.
Bug: webrtc:13741
Change-Id: I7c2cb7789660fd2e082c214d00e50c894666b07c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36406}
* tools_webrtc/PRESUBMIT.py is only checking the licence which is already done here:
38f35db4d4:PRESUBMIT.py;l=913;bpv=1;bpt=0;drc=4fc9bd9f69a0d88889d86d0cc9f8e27406e8a342
* sdk/android/PRESUBMIT.py was added before 'git cl format' was required from the root PRESUBMIT.py:
https://codereview.webrtc.org/2377113003
Bug: webrtc:13895
Change-Id: Ia5ea2529c36ceebfd7d4e6a6a72352bd30c573b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257280
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36391}
Ironic :( The "field trial guy" constructing a invalid string,
if only there would have been a builder instead...
I tested the code several times...but not with debug build...
Bug: webrtc:13741
Change-Id: If3caad0f5533fc150ffd6a34a89ab84f3f0264aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256979
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36370}
This explores the theory that targets that have no files, just
dependencies, are unnecessary.
Bug: webrtc:13805
Change-Id: I1feb50cf3886128031af8970eae361e35fb052c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256974
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36363}
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}
The only caveat is that a name attribute for the `@Parameters`
annotation is required, as otherwise the test infrastructure
doesn’t find test results.
Bug: webrtc:13662
Change-Id: Ib3e2a6671d1045b0e19746ce78dd434fbee78b87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256462
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Auto-Submit: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36322}
This migrates all tests that work by just changing their runner.
This excludes tests using `@RunWith(ParameterizedRunner.class)`, and a
few other non-parameterized tests that fail with the default runner.
Bug: webrtc:13662
Change-Id: Ia0b7c80e04a6a6b7a51348b3a7f587d10061b58e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256367
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36298}
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 in preparation to introduce new java buildtargets that will use the `libaom_av1_encoder` buildtarget instead.
bug: webrtc:13573
Change-Id: I23e80653943ede576657acc17bcc5602cb0a4d5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254540
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36171}
When loading the library fails, the user will be faced with this error:
java.lang.UnsatisfiedLinkError: No implementation found for void org.webrtc.PeerConnectionFactory.nativeInitializeAndroidGlobals()
With no context, however.
Bug: webrtc:13619
Change-Id: I88565f085773ad1e8c2f5742d7fdba96fb6043d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253960
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36150}
WebRTC’s minSdk is 21, so all those checks are dead code.
Change-Id: I26497fd92259b66d9e5ac6afbb393adf4d904c77
Bug: webrtc:13780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253124
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36140}
This change adds a cache for networks in the SimpleNetworkCallback that
is already registered, allowing the cache to be used preferentially as
opposed to the deprecated getAllNetworks call.
This is a fork of https://webrtc-review.googlesource.com/c/src/+/251401
- adds field trials for new behavior
- removes test that did not work
- add (poor) test of field trials
- remove the "network_monitor_java" build target (that I could
not find any reference to...)
Bug: webrtc:13741
Change-Id: I2829c2f1940d4b42455d8e1a2217cf15c133e22b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252284
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36121}
On Android, MediaCodec can request a specific layout of the input buffer.
One can use the stride and slice height to calculate the layout from
the Encoder's MediaFormat. The current code assumes
a specific layout, which is a problematic in Android 12.
Fix this by honoring the stride and slice-height.
Bug: webrtc:13427
Change-Id: I2d3e429309e3add3ae668e0390460b51e6a49eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36033}
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}
These two files turn out to be entangled with each other. Keeping
them together for now.
This is a simpler approach than the one attempted in
https://webrtc-review.googlesource.com/c/src/+/251060
but leaves cleanup of the relationship to a later work item.
Bug: webrtc:13634
Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35942}
Add timestamps to audio_record_jni DataIsRecorded() function, and make
WebRtcAudioRecord find and send the time stamp to that function.
This CL is an continuation of
https://webrtc-review.googlesource.com/c/src/+/249085
Bug: webrtc:13609
Change-Id: I63ab89f1215893cbe1d11d9d8948f5639fc5cdfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249951
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35933}
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>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
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 CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
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}
The default value of sdpSemantics is about to change from PLAN_B to
UNIFIED_PLAN. In order not to cause subtle bugs by applications that
depend on the default value being PLAN_B, we are temporarily making the
default NOT_SPECIFIED. Constructing with NOT_SPECIFIED 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
While we're at it, we're upgrading almost all unit tests to use Unified
Plan. However there are still two tests using Plan B for which I added
TODO comments to be dealt with later; not having an Android setup makes
it impossible to debug these efficiently.
Bug: webrtc:11121
Change-Id: Ib086186bee947d18d31b413e3aeba0cb247b377d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246000
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35700}
This can cause issues on Android S if this initialization happens when
the app doesn't have permission to access the microphone.
Bug: b/197461765
Change-Id: Iebccff9d15f5bb12a7b2c78e1c373e379b37a127
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246104
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35689}
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}
After changing the way libunwind is built in https://crrev.com/c/3297439,
this test should work fine.
Bug: webrtc:13383
Change-Id: I5da7bf27ce3041c934d4ab91367a26c076fac0c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35558}
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}
This is part of the removal of support for SDES.
Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}
It seems the Android CTS tests only verify that 16x16 aligned resolutions
are supported.
This change checks the validity of input frame's size when initialing
or encoding processes are about to start using H/W MediaCodec.
This change has additional APIs to retrieve
|requested_resolution_alignment| and |apply_alignment_to_all_simulcast_layers|
from JAVA VideoEncoder class and its inherited classes. HardwareVideoEncoder
using MediaCodec has values of 16 and true for above variables.
Bug: webrtc:13089
Change-Id: I0c4ebf94eb36da29c2e384a3edf85b82e779b7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229460
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35169}
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}
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".
Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
at org.webrtc.Camera2Session$CaptureSessionCallback.chooseStabilizationMode(Camera2Session.java:234)
at org.webrtc.Camera2Session$CaptureSessionCallback.onConfigured(Camera2Session.java:172)
at android.hardware.camera2.impl.CallbackProxies$SessionStateCallbackProxy.lambda$onConfigured$0(CallbackProxies.java:53)
at android.hardware.camera2.impl.-$$Lambda$CallbackProxies$SessionStateCallbackProxy$soW0qC12Osypoky6AfL3P2-TeDw.run(-.java:4)
at android.os.Handler.handleCallback(Handler.java:873)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:193)
at android.os.HandlerThread.run(HandlerThread.java:65)
Bug: webrtc:13032
Change-Id: Ifb6ef920b700ca03d37c64803c0b34230785846f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227292
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34874}
StreamConfigurationMap.getOutputSizes() may return null:
https://developer.android.com/reference/android/hardware/camera2/params/StreamConfigurationMap#getOutputSizes(java.lang.Class%3CT%3E)
Fixes:
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
at org.webrtc.Camera2Enumerator.convertSizes(Camera2Enumerator.java:234)
at org.webrtc.Camera2Enumerator.getSupportedSizes(Camera2Enumerator.java:147)
at org.webrtc.Camera2Session.findCaptureFormat(Camera2Session.java:325)
at org.webrtc.Camera2Session.start(Camera2Session.java:313)
at org.webrtc.Camera2Session.<init>(Camera2Session.java:296)
at org.webrtc.Camera2Session.create(Camera2Session.java:274)
at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:237)
at android.os.HandlerThread.run(HandlerThread.java:67)
Bug: webrtc:13032
Change-Id: I9154be567cd12c066087818ba22e9cd69e75a22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227291
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34872}
Remove android.support.test.rule.UiThreadTestRule as chromium did in [1] and
Replace android.support.test.annotation.UiThreadTest
with org.chromium.base.test.UiThreadTest.
Also remove unused uiThreadHandler from NetworkMonitorTest.
[1] https://crrev.com/c/2332301
Bug: webrtc:11962
Change-Id: I8f3781d43d4d53d8158c39c81568d8b09b2bec6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230220
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#34864}
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}
Specifically, defer getting the camera index so the error can be
reported instead of crashing:
Fatal Exception: java.lang.IllegalArgumentException: No such camera: Camera 1, Facing front, Orientation 270
at org.webrtc.Camera1Enumerator.getCameraIndex(Camera1Enumerator.java:170)
at org.webrtc.Camera1Capturer.createCameraSession(Camera1Capturer.java:31)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:790)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:214)
at android.os.HandlerThread.run(HandlerThread.java:65)
Bug: webrtc:13032
Change-Id: Ida6bc65046770c11c2b3ee832906e8454cec10df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227290
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34855}
getCameraCharacteristics() may throw IllegalArgumentException:
Fatal Exception: java.lang.IllegalArgumentException: supportsCameraApi:2569: Unknown camera ID 1
at android.hardware.camera2.CameraManager.throwAsPublicException(CameraManager.java:1119)
at android.hardware.camera2.CameraManager.getCameraCharacteristics(CameraManager.java:531)
at org.webrtc.Camera2Session.start(Camera2Session.java:304)
at org.webrtc.Camera2Session.<init>(Camera2Session.java:296)
at org.webrtc.Camera2Session.create(Camera2Session.java:274)
at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:237)
at android.os.HandlerThread.run(HandlerThread.java:67)
Bug: webrtc:13032
Change-Id: I30b6d6da40bc90a94c0c3c79f9dff523182d3da4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227289
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34854}
Fatal Exception: java.lang.NullPointerException: Attempt to read from field 'int org.webrtc.CameraEnumerationAndroid$CaptureFormat.width' on a null object reference
at org.webrtc.Camera2Session$CameraStateCallback.onOpened(Camera2Session.java:122)
at android.hardware.camera2.impl.CameraDeviceImpl$1.run(CameraDeviceImpl.java:151)
at android.os.Handler.handleCallback(Handler.java:938)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:246)
at android.os.HandlerThread.run(HandlerThread.java:67)
Fix NPE when setting the camera2 stabilization mode
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
at org.webrtc.Camera2Session$CaptureSessionCallback.chooseStabilizationMode(Camera2Session.java:234)
at org.webrtc.Camera2Session$CaptureSessionCallback.onConfigured(Camera2Session.java:172)
at android.hardware.camera2.impl.CallbackProxies$SessionStateCallbackProxy.lambda$onConfigured$0(CallbackProxies.java:53)
at android.hardware.camera2.impl.-$$Lambda$CallbackProxies$SessionStateCallbackProxy$soW0qC12Osypoky6AfL3P2-TeDw.run(-.java:4)
at android.os.Handler.handleCallback(Handler.java:873)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:193)
at android.os.HandlerThread.run(HandlerThread.java:65)
Bug: webrtc:13032
Change-Id: I6edd9f0061c445f90ab0881d78183077f89e391f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227294
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34851}
This change does not affect downstream dependencies as androidx.annotation
is fully compatible with android.support.annotation.
Bug: webrtc:11962
Change-Id: I714b473df8d0fee8000ddf3a9beca7c5613db5ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226881
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34839}
PeerConnectionFactory to break off the dependency.
- This is required so that Android app that doesn't use the
peerconnection_java as dependency can include android monitor
directly without incurring size bloat.
Bug: None
Change-Id: I7b3453f268467550c0a4b3a0bbf858d55d2fd8a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229322
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34829}
Before this change HardwareVideoEncoder used capture time as frame
timestamp passed to HW encoder. That led to buffer overshoots with
HW encoders which infer frame rate from timestamps when frames were
dropped before encoding (i.e., frame rate decreases according to frame
timestamps) or when FramerateBitrateAdjuster was used.
Fixed this by using synthetic monotonically increasing timestamps
calculated based on target frame rate provided by bitrate adjuster.
Bug: webrtc:12982
Change-Id: I2454cd4e574bbea1cb9855ced4d998104845415c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228902
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34810}
The simulcast_encoder_adapter expects codecs that specify
supports_native_handle to perform resampling/scaling (through
GetEncoderInfo).
This change adds a method to the RTCVideoEncoder to let objc encoders
specify this rather than relying on the default.
Bug: webrtc:13044
Change-Id: I2efcbd42aa4f2048285f451c7b691fdeca111e62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227641
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34683}
RTCIceCandidate.nativeCandidate returns a unique_ptr that
can be null. As with the previous CL, this is used without checking
whether it is null or not, so it should be fixed.
Bug: None
Change-Id: I70a84f7a2a9a9d47b8cefa198204f9b6d63a7c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227620
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34649}
There are two problems with setLocalDescription / setRemoteDescription
in ObjC SDK.
First, RTCSessionDescription.nativeDescription returns a raw
nullableSessionDescriptionInterface pointer, where sLD/sRD are calling
Clone() method unconditionally, so it might crash.
Second, unnecessary sLD/sRD calls Clone() of the raw pointer and
does not delete it, so this pointer will leak.
To solve these problems, I changed the return type of nativeDescription to
std::unique_ptr and removed the call to Clone() method.
Bug: webrtc:13022, webrtc:13035
Change-Id: Icbb87dda62d3a11af47ec74621cf64b8a6c05228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227380
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/master@{#34647}
This is useful when building the .framework which doesn't need to
export C++ symbols.
Bug: webrtc:12408
Change-Id: Ied775811a72a06b9ad678c9fb549bca286dd7f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227089
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34613}
This allows to get encoder implementation name and other properties
without the need of initializing encoder.
Bug: none
Change-Id: I263a358d562a65a31c420ddb7c4b195316fa5ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226867
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34559}
Need someone from video team devs to be in the list. Working on projects
related to Android media codecs for couple of years and have enough
experience to review the changes. A concrete short-term motivation is
the need to land https://webrtc-review.googlesource.com/c/src/+/226867
Bug: none
Change-Id: I1d0a672f6b497bbe1e2d446386284d568f84664a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226951
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34556}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
Just like the C++ API, add a method in Java VideoFrame.Buffer that
describes the underlying implementation.
Use this method to properly select AndroidVideoBuffer
or AndroidVideoI420Buffer in Java -> C++ Video Frame Conversion.
Also, add a test case for WrappedNativeI420Buffer
in VideoFrameBufferTest for consistency.
Bug: webrtc:12602
Change-Id: I4c0444e8af6f6a1109bc514e7ab6c2214f1f6d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223080
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34545}
This is a step to ensure that the Java to C++ Video Frame Buffer
conversion respects its types.
Bug: webrtc:12602
Change-Id: I1b688b1f421f44474e022b433f9075e75744d86f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223082
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34487}
The deprecation warning started to trigger after the iOS deployment
target has been updated from 10 to 12 by
https://webrtc-review.googlesource.com/c/src/+/224543.
This macro was not defined in tests because the relevant bots were
excluded from CQ when that happened.
Bug: webrtc:12928, webrtc:12937
Change-Id: I6e1891c5080b172cbd74649e0a115b25d6c87d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225020
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34417}
Also changed the logging of exceptions to give more details
Bug: webrtc:10804
Change-Id: Ifba6dee3d1c8ba4ecab408ca7715c3b792d9c004
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222641
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34325}
This method has been deprecated since 2018-07:
https://webrtc-review.googlesource.com/c/src/+/88368/
It is never called by WebRTC itself.
Custom `VideoDecoderFactory` implementations overriding this method can
switch to the overload accepting a `VideoCodecInfo` object.
This is also adding a `toString()` implementation to `VideoCodecInfo`,
to make logging of the value more useful.
Bug: webrtc:7925
Change-Id: I70ec07a0cd4ffa07d165c9851e393439fcc5870b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221960
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34302}
AV1X->AV1 mapping added to SdpVideoFormatToVideoCodecInfo in
https://webrtc-review.googlesource.com/c/src/+/215586 results in
discrepancy of codec name between SDP and VideoCodecInfo. That violates
VideoCodecInfo design and breaks downstream projects.
This CL moves the mapping from VideoCodecInfoToSdpVideoFormat and
SdpVideoFormatToVideoCodecInfo to VideoCodecTypeMime.
Bug: b/181690054
Change-Id: I2a76524c29b082241f2ec72a60a209ce9b0c7c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221205
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34230}
Change-Id: I0880caa77a1097f56c560152e85c9ca29242f825
This PR add support for the `PeerConnectionObserverJni::OnRemoveTrack()`
event on Java, allowing to be notified when a remote track has been
removed. It's a very thing JNI wrapper on top of C++ API, being mostly
similar to other already available events like `track` and `addTrack`.
In Javascript API, tracks are not "removed" explicitly from the
PeerConnection, but instead receiver PeerConnection gets notified that
they have been removed from the streams they are associated to, and when
no `MediaStream` object has that track, it's considered that the track
has been removed from the PeerConnection. In Java and C++ APIs there's no
`MediaStreamObserver` class, so there's no way to listen to the
`removeTrack` event the same way happens in Javascript API, but instead
C++ API has a `removeTrack` event at PeerConnection level. This patchset
just only wraps and expose this `removeTrack` event from the C++ API to
the Java API.
This PR has been sponsored by Atos Research and Innovation
(https://atos.net/en/about-us/innovation-and-research).
Bug: webrtc:12850
Change-Id: I0880caa77a1097f56c560152e85c9ca29242f825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218847
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34225}
With this change, RTCVideoEncoder can specify:
- requested_resolution_alignment,
- apply_alignment_to_all_simulcast_layers
in the same way scaling_settings is specified.
Change-Id: I3de79a2eabaae581d6a9f2ef3e39496b9545a4f5
Bug: webrtc:12829
Change-Id: I3de79a2eabaae581d6a9f2ef3e39496b9545a4f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220933
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Abby Yeh <abbyyeh@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34196}
Deprecate CreateDataChannel, and make it a simple wrapper function.
Bug: webrtc:12796
Change-Id: I053d75a264596ba87ca734a29df9241de93a80c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219784
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34130}
This property doesn't have a getter and it is not required anymore.
Bug: None
Change-Id: Ie3f057cd6928d7fdef4e7971476fb1257900ccc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215261
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34125}
