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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}
142 lines
5 KiB
C++
142 lines
5 KiB
C++
/*
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* Copyright 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_FRAME_TRANSFORMER_INTERFACE_H_
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#define API_FRAME_TRANSFORMER_INTERFACE_H_
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#include <memory>
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#include <string>
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#include "api/ref_count.h"
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#include "api/scoped_refptr.h"
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#include "api/video/encoded_frame.h"
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#include "api/video/video_frame_metadata.h"
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namespace webrtc {
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// Owns the frame payload data.
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class TransformableFrameInterface {
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public:
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virtual ~TransformableFrameInterface() = default;
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// Returns the frame payload data. The data is valid until the next non-const
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// method call.
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virtual rtc::ArrayView<const uint8_t> GetData() const = 0;
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// Copies `data` into the owned frame payload data.
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virtual void SetData(rtc::ArrayView<const uint8_t> data) = 0;
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virtual uint8_t GetPayloadType() const = 0;
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virtual uint32_t GetSsrc() const = 0;
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virtual uint32_t GetTimestamp() const = 0;
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virtual void SetRTPTimestamp(uint32_t timestamp) = 0;
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// TODO(https://bugs.webrtc.org/14878): Change this to pure virtual after it
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// is implemented everywhere.
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virtual absl::optional<Timestamp> GetCaptureTimeIdentifier() const {
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return absl::nullopt;
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}
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enum class Direction {
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kUnknown,
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kReceiver,
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kSender,
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};
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// TODO(crbug.com/1250638): Remove this distinction between receiver and
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// sender frames to allow received frames to be directly re-transmitted on
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// other PeerConnectionss.
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virtual Direction GetDirection() const { return Direction::kUnknown; }
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virtual std::string GetMimeType() const = 0;
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};
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class TransformableVideoFrameInterface : public TransformableFrameInterface {
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public:
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virtual ~TransformableVideoFrameInterface() = default;
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virtual bool IsKeyFrame() const = 0;
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virtual VideoFrameMetadata Metadata() const = 0;
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virtual void SetMetadata(const VideoFrameMetadata&) = 0;
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};
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// Extends the TransformableFrameInterface to expose audio-specific information.
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class TransformableAudioFrameInterface : public TransformableFrameInterface {
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public:
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virtual ~TransformableAudioFrameInterface() = default;
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virtual rtc::ArrayView<const uint32_t> GetContributingSources() const = 0;
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virtual const absl::optional<uint16_t> SequenceNumber() const = 0;
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virtual absl::optional<uint64_t> AbsoluteCaptureTimestamp() const = 0;
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enum class FrameType { kEmptyFrame, kAudioFrameSpeech, kAudioFrameCN };
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// TODO(crbug.com/1456628): Change this to pure virtual after it
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// is implemented everywhere.
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virtual FrameType Type() const { return FrameType::kEmptyFrame; }
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// Audio level in -dBov. Values range from 0 to 127, representing 0 to -127
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// dBov. 127 represents digital silence. Only present on remote frames if
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// the audio level header extension was included.
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virtual absl::optional<uint8_t> AudioLevel() const = 0;
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};
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// Objects implement this interface to be notified with the transformed frame.
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class TransformedFrameCallback : public rtc::RefCountInterface {
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public:
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virtual void OnTransformedFrame(
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std::unique_ptr<TransformableFrameInterface> frame) = 0;
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// Request to no longer be called on each frame, instead having frames be
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// sent directly to OnTransformedFrame without additional work.
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// TODO(crbug.com/1502781): Make pure virtual once all mocks have
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// implementations.
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virtual void StartShortCircuiting() {}
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protected:
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~TransformedFrameCallback() override = default;
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};
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// Transforms encoded frames. The transformed frame is sent in a callback using
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// the TransformedFrameCallback interface (see above).
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class FrameTransformerInterface : public rtc::RefCountInterface {
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public:
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// Transforms `frame` using the implementing class' processing logic.
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virtual void Transform(
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std::unique_ptr<TransformableFrameInterface> transformable_frame) = 0;
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virtual void RegisterTransformedFrameCallback(
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rtc::scoped_refptr<TransformedFrameCallback>) {}
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virtual void RegisterTransformedFrameSinkCallback(
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rtc::scoped_refptr<TransformedFrameCallback>,
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uint32_t ssrc) {}
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virtual void UnregisterTransformedFrameCallback() {}
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virtual void UnregisterTransformedFrameSinkCallback(uint32_t ssrc) {}
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protected:
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~FrameTransformerInterface() override = default;
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};
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// An interface implemented by classes that can host a transform.
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// Currently this is implemented by the RTCRtpSender and RTCRtpReceiver.
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class FrameTransformerHost {
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public:
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virtual ~FrameTransformerHost() {}
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virtual void SetFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
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// TODO: bugs.webrtc.org/15929 - To be added:
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// virtual AddIncomingMediaType(RtpCodec codec) = 0;
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// virtual AddOutgoingMediaType(RtpCodec codec) = 0;
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};
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} // namespace webrtc
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#endif // API_FRAME_TRANSFORMER_INTERFACE_H_
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