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}
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}
This change replaces type of absolute_capture_timestamp_ms_ in
TransformableOutgoingAudioFrame from int to optional uint and makes
the function AbsoluteCaptureTimestamp() inside
TransformableAudioFrameInterface pure virtual.
Bug: webrtc:14949
Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40814}
Keep the logic managing whether audio RTP timestamps have the random
start offset added or not inside ChannelSend, so that the
ChannelSendFrameTransformerDelegate doesn't need to worry about it.
Crucially, this means that frames moved between senders by encoded
transforms clients will always use the correct offset for the channel
where we actually get sent.
Also rename TS variables throughout both classes to be explicit over
whether the offset has been added or not.
Bug: chromium:1464847
Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40655}
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}
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 will clone an encoded audio frame into a sender frame.
Bug: webrtc:14949
Change-Id: Ie62d9f5ec457541b335bde8f2f6e9b6d24704cf6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294560
Commit-Queue: Tove Petersson <tovep@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39480}
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}
includes the (random) rtp start offset in the timestamp passed to the frame transformer callback
Bug: chromium:1069278
Change-Id: I7d10130404d93df7cee3b8f87a0b780801a51415
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173329
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31067}
