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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_TEST_RTPFILE_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
     13 
     14 #include <stdio.h>
     15 #include <queue>
     16 
     17 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
     18 #include "webrtc/modules/include/module_common_types.h"
     19 #include "webrtc/system_wrappers/include/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 size_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 size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
     36                       size_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, size_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   size_t payloadSize;
     59   uint32_t frequency;
     60 };
     61 
     62 class RTPBuffer : public RTPStream {
     63  public:
     64   RTPBuffer();
     65 
     66   ~RTPBuffer();
     67 
     68   void Write(const uint8_t payloadType,
     69              const uint32_t timeStamp,
     70              const int16_t seqNo,
     71              const uint8_t* payloadData,
     72              const size_t payloadSize,
     73              uint32_t frequency) override;
     74 
     75   size_t Read(WebRtcRTPHeader* rtpInfo,
     76               uint8_t* payloadData,
     77               size_t payloadSize,
     78               uint32_t* offset) override;
     79 
     80   bool EndOfFile() const override;
     81 
     82  private:
     83   RWLockWrapper* _queueRWLock;
     84   std::queue<RTPPacket *> _rtpQueue;
     85 };
     86 
     87 class RTPFile : public RTPStream {
     88  public:
     89   ~RTPFile() {
     90   }
     91 
     92   RTPFile()
     93       : _rtpFile(NULL),
     94         _rtpEOF(false) {
     95   }
     96 
     97   void Open(const char *outFilename, const char *mode);
     98 
     99   void Close();
    100 
    101   void WriteHeader();
    102 
    103   void ReadHeader();
    104 
    105   void Write(const uint8_t payloadType,
    106              const uint32_t timeStamp,
    107              const int16_t seqNo,
    108              const uint8_t* payloadData,
    109              const size_t payloadSize,
    110              uint32_t frequency) override;
    111 
    112   size_t Read(WebRtcRTPHeader* rtpInfo,
    113               uint8_t* payloadData,
    114               size_t payloadSize,
    115               uint32_t* offset) override;
    116 
    117   bool EndOfFile() const override { return _rtpEOF; }
    118 
    119  private:
    120   FILE* _rtpFile;
    121   bool _rtpEOF;
    122 };
    123 
    124 }  // namespace webrtc
    125 
    126 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
    127