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      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_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
     13 
     14 #include <stdio.h>
     15 #include <queue>
     16 
     17 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
     18 #include "webrtc/modules/interface/module_common_types.h"
     19 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
     20 #include "webrtc/typedefs.h"
     21 
     22 namespace webrtc {
     23 
     24 class RTPStream {
     25  public:
     26   virtual ~RTPStream() {
     27   }
     28 
     29   virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
     30                      const int16_t seqNo, const uint8_t* payloadData,
     31                      const uint16_t payloadSize, uint32_t frequency) = 0;
     32 
     33   // Returns the packet's payload size. Zero should be treated as an
     34   // end-of-stream (in the case that EndOfFile() is true) or an error.
     35   virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
     36                         uint16_t payloadSize, uint32_t* offset) = 0;
     37   virtual bool EndOfFile() const = 0;
     38 
     39  protected:
     40   void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
     41                      uint32_t timeStamp, uint32_t ssrc);
     42 
     43   void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
     44 };
     45 
     46 class RTPPacket {
     47  public:
     48   RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
     49             const uint8_t* payloadData, uint16_t payloadSize,
     50             uint32_t frequency);
     51 
     52   ~RTPPacket();
     53 
     54   uint8_t payloadType;
     55   uint32_t timeStamp;
     56   int16_t seqNo;
     57   uint8_t* payloadData;
     58   uint16_t payloadSize;
     59   uint32_t frequency;
     60 };
     61 
     62 class RTPBuffer : public RTPStream {
     63  public:
     64   RTPBuffer();
     65 
     66   ~RTPBuffer();
     67 
     68   virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
     69                      const int16_t seqNo, const uint8_t* payloadData,
     70                      const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
     71 
     72   virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
     73                         uint16_t payloadSize, uint32_t* offset) OVERRIDE;
     74 
     75   virtual bool EndOfFile() const OVERRIDE;
     76 
     77  private:
     78   RWLockWrapper* _queueRWLock;
     79   std::queue<RTPPacket *> _rtpQueue;
     80 };
     81 
     82 class RTPFile : public RTPStream {
     83  public:
     84   ~RTPFile() {
     85   }
     86 
     87   RTPFile()
     88       : _rtpFile(NULL),
     89         _rtpEOF(false) {
     90   }
     91 
     92   void Open(const char *outFilename, const char *mode);
     93 
     94   void Close();
     95 
     96   void WriteHeader();
     97 
     98   void ReadHeader();
     99 
    100   virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
    101                      const int16_t seqNo, const uint8_t* payloadData,
    102                      const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
    103 
    104   virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
    105                         uint16_t payloadSize, uint32_t* offset) OVERRIDE;
    106 
    107   virtual bool EndOfFile() const OVERRIDE {
    108     return _rtpEOF;
    109   }
    110 
    111  private:
    112   FILE* _rtpFile;
    113   bool _rtpEOF;
    114 };
    115 
    116 }  // namespace webrtc
    117 
    118 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
    119