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// Copyright 2017 The Chromium Authors. All rights reserved.
// 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_PLATFORM_PEERCONNECTION_RTC_RTP_RECEIVER_PLATFORM_H_
#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_RTP_RECEIVER_PLATFORM_H_
#include <memory>
#include "base/optional.h"
#include "third_party/blink/renderer/platform/peerconnection/rtc_stats.h"
#include "third_party/blink/renderer/platform/platform_export.h"
#include "third_party/blink/renderer/platform/wtf/text/wtf_string.h"
#include "third_party/blink/renderer/platform/wtf/vector.h"
#include "third_party/webrtc/api/dtls_transport_interface.h"
#include "third_party/webrtc/api/rtp_parameters.h"
#include "third_party/webrtc/api/stats/rtc_stats.h"
namespace blink {
class RTCEncodedAudioStreamTransformer;
class RTCEncodedVideoStreamTransformer;
class RTCRtpSource;
class MediaStreamComponent;
// Implementations of this interface keep the corresponding WebRTC-layer
// receiver alive through reference counting. Multiple |RTCRtpReceiverPlatform|s
// could reference the same receiver, see |id|.
// https://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
class PLATFORM_EXPORT RTCRtpReceiverPlatform {
public:
virtual ~RTCRtpReceiverPlatform();
virtual std::unique_ptr<RTCRtpReceiverPlatform> ShallowCopy() const = 0;
// Two |RTCRtpReceiverPlatform|s referencing the same WebRTC-layer receiver
// have the same |id|.
virtual uintptr_t Id() const = 0;
virtual rtc::scoped_refptr<webrtc::DtlsTransportInterface>
DtlsTransport() = 0;
// Note: For convenience, DtlsTransportInformation always returns a value.
// The information is only interesting if DtlsTransport() is non-null.
virtual webrtc::DtlsTransportInformation DtlsTransportInformation() = 0;
virtual MediaStreamComponent* Track() const = 0;
virtual Vector<String> StreamIds() const = 0;
virtual Vector<std::unique_ptr<RTCRtpSource>> GetSources() = 0;
virtual void GetStats(RTCStatsReportCallback,
const Vector<webrtc::NonStandardGroupId>&) = 0;
virtual std::unique_ptr<webrtc::RtpParameters> GetParameters() const = 0;
virtual void SetJitterBufferMinimumDelay(
base::Optional<double> delay_seconds) = 0;
virtual RTCEncodedAudioStreamTransformer* GetEncodedAudioStreamTransformer()
const {
return nullptr;
}
virtual RTCEncodedVideoStreamTransformer* GetEncodedVideoStreamTransformer()
const {
return nullptr;
}
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_PLATFORM_PEERCONNECTION_RTC_RTP_RECEIVER_PLATFORM_H_
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