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// Copyright 2020 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifndef THIRD_PARTY_BLINK_RENDERER_MODULES_PEERCONNECTION_RTC_ENCODED_AUDIO_FRAME_DELEGATE_H_
#define THIRD_PARTY_BLINK_RENDERER_MODULES_PEERCONNECTION_RTC_ENCODED_AUDIO_FRAME_DELEGATE_H_
#include <stdint.h>
#include <memory>
#include "base/synchronization/lock.h"
#include "base/thread_annotations.h"
#include "base/time/time.h"
#include "base/types/expected.h"
#include "third_party/blink/renderer/bindings/core/v8/serialization/serialized_script_value.h"
#include "third_party/blink/renderer/bindings/modules/v8/v8_rtc_encoded_audio_frame_metadata.h"
#include "third_party/blink/renderer/core/execution_context/execution_context.h"
#include "third_party/blink/renderer/modules/modules_export.h"
#include "third_party/blink/renderer/modules/peerconnection/peer_connection_util.h"
#include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
#include "third_party/webrtc/api/frame_transformer_interface.h"
#include "third_party/webrtc/api/units/timestamp.h"
namespace blink {
class DOMArrayBuffer;
// This class wraps a WebRTC audio frame and allows making shallow
// copies. Its purpose is to support making RTCEncodedVideoFrames
// serializable in the same process.
class RTCEncodedAudioFrameDelegate
: public ThreadSafeRefCounted<RTCEncodedAudioFrameDelegate> {
public:
explicit RTCEncodedAudioFrameDelegate(
std::unique_ptr<webrtc::TransformableAudioFrameInterface> webrtc_frame,
webrtc::ArrayView<const unsigned int> contributing_sources,
std::optional<uint16_t> sequence_number);
uint32_t RtpTimestamp() const;
DOMArrayBuffer* CreateDataBuffer(v8::Isolate* isolate) const;
void SetData(const DOMArrayBuffer* data);
// This method can only be called from the main thread.
base::expected<void, String> SetWebRtcFrameMetadata(
ExecutionContext*,
const RTCEncodedAudioFrameMetadata*);
std::optional<uint32_t> Ssrc() const;
std::optional<uint8_t> PayloadType() const;
std::optional<std::string> MimeType() const;
std::optional<uint16_t> SequenceNumber() const;
Vector<uint32_t> ContributingSources() const;
std::optional<base::TimeTicks> ReceiveTime() const;
std::optional<CaptureTimeInfo> CaptureTime() const;
std::optional<base::TimeDelta> SenderCaptureTimeOffset() const;
std::optional<double> AudioLevel() const;
std::unique_ptr<webrtc::TransformableAudioFrameInterface> PassWebRtcFrame();
std::unique_ptr<webrtc::TransformableAudioFrameInterface> CloneWebRtcFrame();
private:
std::optional<base::TimeTicks> ComputeReceiveTime() const
EXCLUSIVE_LOCKS_REQUIRED(&lock_);
std::optional<CaptureTimeInfo> ComputeCaptureTime() const
EXCLUSIVE_LOCKS_REQUIRED(&lock_);
std::optional<base::TimeDelta> ComputeSenderCaptureTimeOffset() const
EXCLUSIVE_LOCKS_REQUIRED(&lock_);
std::optional<double> ComputeAudioLevel() const
EXCLUSIVE_LOCKS_REQUIRED(&lock_);
mutable base::Lock lock_;
std::unique_ptr<webrtc::TransformableAudioFrameInterface> webrtc_frame_
GUARDED_BY(lock_);
const Vector<uint32_t> contributing_sources_;
const std::optional<uint16_t> sequence_number_;
struct Metadata {
std::optional<uint32_t> ssrc;
std::optional<uint8_t> payload_type;
std::optional<std::string> mime_type;
std::optional<base::TimeTicks> receive_time;
std::optional<CaptureTimeInfo> capture_time_info;
std::optional<base::TimeDelta> sender_capture_time_offset;
std::optional<double> audio_level;
uint32_t rtp_timestamp = 0;
};
// This field is set after the frame is neutered (e.g., written to a stream or
// transferred).
Metadata post_neuter_metadata_ GUARDED_BY(lock_);
};
class MODULES_EXPORT RTCEncodedAudioFramesAttachment
: public SerializedScriptValue::Attachment {
public:
static const void* kAttachmentKey;
RTCEncodedAudioFramesAttachment() = default;
~RTCEncodedAudioFramesAttachment() override = default;
bool IsLockedToAgentCluster() const override {
return !encoded_audio_frames_.empty();
}
Vector<scoped_refptr<RTCEncodedAudioFrameDelegate>>& EncodedAudioFrames() {
return encoded_audio_frames_;
}
const Vector<scoped_refptr<RTCEncodedAudioFrameDelegate>>&
EncodedAudioFrames() const {
return encoded_audio_frames_;
}
private:
Vector<scoped_refptr<RTCEncodedAudioFrameDelegate>> encoded_audio_frames_;
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_MODULES_PEERCONNECTION_RTC_ENCODED_AUDIO_FRAME_DELEGATE_H_
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