Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
      12             : #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
      13             : 
      14             : #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
      15             : #include "webrtc/typedefs.h"
      16             : 
      17             : namespace webrtc {
      18             : 
      19             : struct CodecInst;
      20             : class RTPPayloadRegistry;
      21             : class VideoCodec;
      22             : 
      23           0 : class TelephoneEventHandler {
      24             :  public:
      25           0 :   virtual ~TelephoneEventHandler() {}
      26             : 
      27             :   // The following three methods implement the TelephoneEventHandler interface.
      28             :   // Forward DTMFs to decoder for playout.
      29             :   virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
      30             : 
      31             :   // Is forwarding of outband telephone events turned on/off?
      32             :   virtual bool TelephoneEventForwardToDecoder() const = 0;
      33             : 
      34             :   // Is TelephoneEvent configured with payload type payload_type
      35             :   virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
      36             : };
      37             : 
      38           0 : class RtpReceiver {
      39             :  public:
      40             :   // Creates a video-enabled RTP receiver.
      41             :   static RtpReceiver* CreateVideoReceiver(
      42             :       Clock* clock,
      43             :       RtpData* incoming_payload_callback,
      44             :       RtpFeedback* incoming_messages_callback,
      45             :       RTPPayloadRegistry* rtp_payload_registry);
      46             : 
      47             :   // Creates an audio-enabled RTP receiver.
      48             :   static RtpReceiver* CreateAudioReceiver(
      49             :       Clock* clock,
      50             :       RtpData* incoming_payload_callback,
      51             :       RtpFeedback* incoming_messages_callback,
      52             :       RTPPayloadRegistry* rtp_payload_registry);
      53             : 
      54           0 :   virtual ~RtpReceiver() {}
      55             : 
      56             :   // Returns a TelephoneEventHandler if available.
      57             :   virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
      58             : 
      59             :   // Registers a receive payload in the payload registry and notifies the media
      60             :   // receiver strategy.
      61             :   virtual int32_t RegisterReceivePayload(const CodecInst& audio_codec) = 0;
      62             :   // Registers a receive payload in the payload registry.
      63             :   virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec) = 0;
      64             : 
      65             :   // De-registers |payload_type| from the payload registry.
      66             :   virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
      67             : 
      68             :   // Parses the media specific parts of an RTP packet and updates the receiver
      69             :   // state. This for instance means that any changes in SSRC and payload type is
      70             :   // detected and acted upon.
      71             :   virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
      72             :                                  const uint8_t* payload,
      73             :                                  size_t payload_length,
      74             :                                  PayloadUnion payload_specific,
      75             :                                  bool in_order) = 0;
      76             : 
      77             :   // Gets the last received timestamp. Returns true if a packet has been
      78             :   // received, false otherwise.
      79             :   virtual bool Timestamp(uint32_t* timestamp) const = 0;
      80             :   // Gets the time in milliseconds when the last timestamp was received.
      81             :   // Returns true if a packet has been received, false otherwise.
      82             :   virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
      83             : 
      84             :   // Returns the remote SSRC of the currently received RTP stream.
      85             :   virtual uint32_t SSRC() const = 0;
      86             : 
      87             :   // Returns the current remote CSRCs.
      88             :   virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
      89             :  
      90             :   virtual void GetRID(char rid[256]) const = 0;
      91             : 
      92             :   // Returns the current energy of the RTP stream received.
      93             :   virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
      94             : };
      95             : }  // namespace webrtc
      96             : 
      97             : #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_