|  | /* | 
|  | *  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 { | 
|  |  | 
|  | 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( | 
|  | int payload_type, | 
|  | const SdpAudioFormat& audio_format) = 0; | 
|  |  | 
|  | // Deprecated version of the above. | 
|  | int32_t RegisterReceivePayload(const CodecInst& audio_codec); | 
|  |  | 
|  | // 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) = 0; | 
|  | // TODO(nisse): Deprecated version, delete as soon as downstream | 
|  | // applications are updated. | 
|  | bool IncomingRtpPacket(const RTPHeader& rtp_header, | 
|  | const uint8_t* payload, | 
|  | size_t payload_length, | 
|  | PayloadUnion payload_specific, | 
|  | bool in_order /* Ignored */) { | 
|  | return IncomingRtpPacket(rtp_header, payload, payload_length, | 
|  | payload_specific); | 
|  | } | 
|  |  | 
|  | // Gets the RTP timestamp and the corresponding monotonic system | 
|  | // time for the most recent in-order packet. Returns true on | 
|  | // success, false if no packet has been received. | 
|  | virtual bool GetLatestTimestamps(uint32_t* timestamp, | 
|  | 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_ |