Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
This makes SetChannel() consistently make 2 invokes instead of a
multiple of senders+receivers (previous minimum was 4 but could be
larger).
* Stop() doesn't hop to the worker thread.
* SetMediaChannel(), an already-required step on the worker thread for
senders and *sometimes* for receivers[1], is now consistently required
for both. This simplifies transceiver teardown and enables the next
bullet.
* Transceiver stops all senders and receivers in one go rather than
ping ponging between threads.
[1] When not required, it was done implicitly inside of Stop().
See changes in `RtpTransceiver::SetChannel`
Bug: webrtc:13540
Change-Id: Ied61636c8ef09d782bf519524fff2a31e15219a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249797
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36057}
This is a reland of 3ed36c0521546881656c73984456485dcab16205
Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}
Bug: webrtc:13540
Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35967}
This reverts commit 3ed36c0521546881656c73984456485dcab16205.
Reason for revert: Breaks downstream project.
Original change's description:
> Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver."
>
> This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
>
> The difference from the original CL is that a check for
> `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
> This caused a side effect that registering the sink while the source
> was in an "initializing" state, failed. The last remaining state
> however, is `kEnded` - but since there's no logic in the class around
> the expected value of the states, the check inside of AddSink()
> doesn't provide an additional value - it's rather a surprise for
> developers if it doesn't succeed. So, now removed.
>
> Original change's description:
> > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
> >
> > This simplifies the logic in these classes a bit, which makes upcoming
> > change easier. The `stopped_` flag in these classes was essentially
> > the same thing as `media_channel_ == nullptr`, which is what's
> > consistently used now for the same checks.
> >
> > Bug: webrtc:13540
> > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35907}
>
> Bug: webrtc:13540
> Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
> Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35958}
TBR=ilnik@webrtc.org,tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Ieb7235d88c808c78ad0847403be991d4dce1ace6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:13540
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251383
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@{#35963}
This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f
The difference from the original CL is that a check for
`state_ == kLive` inside of RemoteAudioSource::AddSink has been removed.
This caused a side effect that registering the sink while the source
was in an "initializing" state, failed. The last remaining state
however, is `kEnded` - but since there's no logic in the class around
the expected value of the states, the check inside of AddSink()
doesn't provide an additional value - it's rather a surprise for
developers if it doesn't succeed. So, now removed.
Original change's description:
> Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
>
> This simplifies the logic in these classes a bit, which makes upcoming
> change easier. The `stopped_` flag in these classes was essentially
> the same thing as `media_channel_ == nullptr`, which is what's
> consistently used now for the same checks.
>
> Bug: webrtc:13540
> Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35907}
Bug: webrtc:13540
Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35958}
This reverts commit bb57e2d7aa9b36843233d1394422f03d12d9c31f.
Reason for revert: Speculative revert to see if this is causing
breakage in Chromium
Bug: chromium:13665
Original change's description:
> Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver.
>
> This simplifies the logic in these classes a bit, which makes upcoming
> change easier. The `stopped_` flag in these classes was essentially
> the same thing as `media_channel_ == nullptr`, which is what's
> consistently used now for the same checks.
>
> Bug: webrtc:13540
> Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35907}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13540
Change-Id: I67fb2c26b6931b80e3aab749443122d62a82855d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251141
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35938}
This simplifies the logic in these classes a bit, which makes upcoming
change easier. The `stopped_` flag in these classes was essentially
the same thing as `media_channel_ == nullptr`, which is what's
consistently used now for the same checks.
Bug: webrtc:13540
Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35907}
For implementations where the signaling and worker threads are not
the same thread, this significantly cuts down on Thread::Invoke()s that
would block the signaling thread while waiting for the worker thread.
For Audio and Video Rtp receivers, the following methods now do not
block the signaling thread:
* GetParameters
* SetJitterBufferMinimumDelay
* GetSources
* SetFrameDecryptor / GetFrameDecryptor
* SetDepacketizerToDecoderFrameTransformer
Importantly this change also makes the track() accessor accessible
directly from the application thread (bypassing the proxy) since
for receiver objects, the track object is const.
Other changes:
* Remove RefCountedObject inheritance, use make_ref_counted instead.
* Every member variable in the rtp receiver classes is now RTC_GUARDED
* Stop() now fully clears up worker thread state, and Stop() is
consistently called before destruction. This means that there's one
thread hop instead of at least 4 before (sometimes more), per receiver.
* OnChanged triggered volume for audio tracks is done asynchronously.
* Deleted most of the JitterBufferDelay implementation. Turns out that
it was largely unnecessary overhead and complexity.
It seems that these two classes are copy/pasted to a large extent
so further refactoring would be good in the future, as to not have to
fix each issue twice.
Bug: chromium:1184611
Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34022}
This CL mostly adds plumbing to get awareness of the network thread
to the media channel classes. Currently this pointer is only used
to DCHECK that `SetInterface` for the `NetworkInterface` pointer, is
called on the network thread. Follow up changes will establish that
most of the methods are called on the network thread and the mutex
in the MediaChannel base class, can be removed.
Most of the changes in the CL are in channel_unittest.cc. They're mostly
around updating the tests to incorporate the network thread in ways
that reflect how the classes are used in production. Another change is
to use accessor methods for the media channel instances instead of
caching potentially dangling pointers.
Bug: webrtc:11993
Change-Id: I8e2ed1bc23724e238554dbce386789d69660f7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217682
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33951}
This change finally wires up VideoRtpReceiver::OnGenerateKeyFrame and
OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.
Bug: chromium:1013590
Change-Id: I76f8226752294aee8fe137d1a78ee66548900cc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161095
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30003}
