Commit graph

9 commits

Author SHA1 Message Date
Tony Herre
9c6874607a Consolidate encoded transform mocks into api/test/
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/

Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
2024-01-26 12:46:34 +00:00
Tony Herre
6e956053b7 Support shortcircuiting encoded transforms
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}
2023-11-17 13:03:27 +00:00
Philipp Hancke
d2098933e1 Expose audio mimeType for insertable streams
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}
2023-11-03 09:49:12 +00:00
Palak Agarwal
14e5d4ce5e Support sending IncomingFrames in audio
ChannelSendFrameTransformerDelegate::SendFrame() currently only
supports sending frames in a single direction. With this change, we
allow sending received audio frames.

Bug: chromium:1464847
Change-Id: I8113a3278dfce7b2ba709afecc672bc9af9c4a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316600
Reviewed-by: 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@{#40643}
2023-08-28 13:52:20 +00:00
Palak Agarwal
f263e1eb03 Support receiving cloned encoded audio frames
Bug: chromium:1464860
Change-Id: I01b2d768fcf5aef09b32304a8f9fe0b00ca32357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316320
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@{#40583}
2023-08-21 09:43:52 +00:00
Niels Möller
105711e9ad Move rtc::make_ref_counted to api/
Bug: webrtc:12701
Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390
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@{#37219}
2022-06-15 09:47:38 +00:00
Niels Möller
83830f316e Delete TestListener and top-level thread wrapping.
Instead use rtc::AutoThread in tests that need that.

Bug: webrtc:9714
Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36950}
2022-05-20 15:21:21 +00:00
Tomas Gunnarsson
c1d589146b Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
Bug: webrtc:12701
Change-Id: Ie50225374f811424faf20caf4cf454b2fd1c4dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215930
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33818}
2021-04-23 12:04:39 +00:00
Marina Ciocea
701ccf97c9 Add unit tests for audio receive channel frame transformer delegate.
Bug: webrtc:11380
Change-Id: I4630b75c83886d722e7be64d50a9790c20956ba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174004
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31157}
2020-05-04 15:44:08 +00:00