and files that broke when I fixed the first set.
Bug: webrtc:42226242
Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42429}
This makes the downcasts currently used in eg
modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
safer.
Bug: webrtc:339815768
Change-Id: Ie6c7ab84666d399dbca4c95846b850aac5327f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42325}
Calculate the RMS audio level of audio packets being sent before
invoking an encoded frame transform, and pass them with the encode frame
object.
Before this, the audio level was calculated at send time by having rms_levels_ look at all audio samples encoded since the last send. This
is fine without a transform, as this is done synchronously after
encoding, but with an async transform which might take arbitrarily long,
we could end up marking older audio packets with newer audio levels, or
not at all.
This also makes things work correctly if external encoded frames are
injected from elsewhere to be sent, and exposes the AudioLevel on the
TransformableFrame interface.
Bug: chromium:337193823, webrtc:42226202
Change-Id: If55d2c1d30dc03408ca9fb0193d791db44428316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349263
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42193}
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.
The old fields are preserved for compatibility with downstream projects, but will be removed in the future.
Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
Failing a DCHECK on a ChannelReceiver having its encoded transform set
more than once contradicts the comment above - this can happen when
reconfiguring a channel, eg as in the web platform test
webrtc/recvonly-transceiver-can-become-sendrecv.https.html.
It was added after the original code by a different author, and indeed
the video side doesn't have such a check.
Bug: chromium:1502781
Change-Id: Id36e52601da34ebc194ff058e4672046379f576e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328560
Commit-Queue: Tony Herre <herre@google.com>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41246}
Add a StartShortCircuiting() callback to allow clients which have
configured Encoded Transforms when creating a PeerConnection to have
all frames skip the transform. This offers a zero cost path for streams
which don't need transforms.
This is preferable to uninstalling/not installing the transform to allow
implementing the behaviour in
https://w3c.github.io/webrtc-encoded-transform/#stream-creation -
giving web apps a chance to configure transforms within a short window
(before the next JS event loop run, so usually sub-millisecond) after stream creation, without any untransformed frames passing.
Usage in Chromium: crrev.com/c/5040731
Bug: chromium:1502781
Change-Id: I803477db1df51e80bdedf6c84d2d3695b088de83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41184}
Split from
https://webrtc-review.googlesource.com/c/src/+/318283
to reduce CL size. Takes a different and (hopefully) simpler
approach.
BUG=webrtc:15579
Change-Id: I8517ffbeb0f0a76db80e3e367de727fb6976211d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325023
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41073}
Instead switch to specific getters, or methods only defined on specific implementations rather than part of the public API.
Once uses are removed from Chromium, I'll mark GetHeader() deprecated
and eventually remove it.
Bug: chromium:1456628
Change-Id: I19b80489b3a0322c201e24994494cfbb742ee13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309780
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40344}
Make outgoing encoded audio frames inherit from the same Audio interface
that incoming frames inherit from, to align them and make it possible to
eg clone frames regardless of their direction.
Also begin removing GetHeader() from the Audio interface, replacing it
with getters for the specific values we actually need to propagate in
the API: sequence number and CSRCs. This makes it much easier to treat
incoming and outgoing frames the same, even if they don't have full
RtpHeaders prepared at the point of the transform.
Bug: chromium:1453226
Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40309}
This change will make it possible to let us modify timestamp in
RTCEncodedAudioFrame.
Change-Id: I97e9571c258fd718d6c211014f1476ca46c78097
Bug: webrtc:14709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307501
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40238}
RTPHeader is not exported, so the TransformableIncomingAudioFrame can't be mocked in chrome tests, using a getter instead.
Bug: chromium:1247260
Change-Id: I2af4e6a88b3f4772b3bb50ee0ae9d5c80fed3ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278785
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38352}
Currently the implementation of FrameTransformers uses distinct,
incompatible types for recevied vs about-to-be-sent frames. This adds a
flag in the interface so we can at least check that we are being given
the correct type. crbug.com/1250638 tracks removing the need for this.
Chrome will be updated after this to check the direction flag and provide
a javascript error if the wrong type of frame is written into the
encoded insertable streams writable stream, rather than crashing.
Bug: chromium:1247260
Change-Id: I9cbb66962ea0718ed47c5e5dba19a8ff9635b0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <toprice@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35100}
* Removes playing_lock_, sync_info_lock_ and video_sync_lock_.
* Also remove video_capture_thread_race_checker_ which was redundant.
Only video_sync_lock_ was actually needed. The other two aren't needed
anymore because of changes made to RtpStreamsSynchronizer class last
year (see webrtc:11489).
In the one case where we had a lock, we post a task to the thread
where the state is maintained. This task is for capturing histograms
which I'm not sure we should have been capturing on the audio thread
anyway.
Also making ChannelReceiveFrameTransformerDelegate compatible with more
tests by using TaskQueueBase instead of rtc::Thread. A number of tests
that instantiate ChannelReceive (and thereby CRFTD) set the worker
thread as a TQ and not actually an rtc::Thread instance. In those cases
CRFTD would previously have gotten a nullptr for the worker thread.
Bug: webrtc:11993
Change-Id: I59f4b2afbfedb06f241d9a613f8538adc19cd6d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221364
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34257}
