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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
#define MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
#include <vector>
#include "api/rtpreceiverinterface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
struct CodecInst;
class RTPPayloadRegistry;
class VideoCodec;
class TelephoneEventHandler {
public:
virtual ~TelephoneEventHandler() {}
// The following three methods implement the TelephoneEventHandler interface.
// Forward DTMFs to decoder for playout.
virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
// Is forwarding of outband telephone events turned on/off?
virtual bool TelephoneEventForwardToDecoder() const = 0;
// Is TelephoneEvent configured with payload type payload_type
virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
};
class RtpReceiver {
public:
// Creates a video-enabled RTP receiver.
static RtpReceiver* CreateVideoReceiver(
Clock* clock,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry);
// Creates an audio-enabled RTP receiver.
static RtpReceiver* CreateAudioReceiver(
Clock* clock,
RtpData* incoming_payload_callback,
RtpFeedback* incoming_messages_callback,
RTPPayloadRegistry* rtp_payload_registry);
virtual ~RtpReceiver() {}
// Returns a TelephoneEventHandler if available.
virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
// Registers a receive payload in the payload registry and notifies the media
// receiver strategy.
virtual int32_t RegisterReceivePayload(const CodecInst& audio_codec) = 0;
// Registers a receive payload in the payload registry.
virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec) = 0;
// De-registers |payload_type| from the payload registry.
virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
// Parses the media specific parts of an RTP packet and updates the receiver
// state. This for instance means that any changes in SSRC and payload type is
// detected and acted upon.
virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
const uint8_t* payload,
size_t payload_length,
PayloadUnion payload_specific,
bool in_order) = 0;
// Gets the last received timestamp. Returns true if a packet has been
// received, false otherwise.
virtual bool Timestamp(uint32_t* timestamp) const = 0;
// Gets the time in milliseconds when the last timestamp was received.
// Returns true if a packet has been received, false otherwise.
virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
// Returns the remote SSRC of the currently received RTP stream.
virtual uint32_t SSRC() const = 0;
// Returns the current remote CSRCs.
virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
// Returns the current energy of the RTP stream received.
virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
virtual std::vector<RtpSource> GetSources() const = 0;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_