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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_ENCODEDECODETEST_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
     13 
     14 #include <stdio.h>
     15 #include <string.h>
     16 
     17 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
     18 #include "webrtc/modules/audio_coding/test/ACMTest.h"
     19 #include "webrtc/modules/audio_coding/test/PCMFile.h"
     20 #include "webrtc/modules/audio_coding/test/RTPFile.h"
     21 #include "webrtc/typedefs.h"
     22 
     23 namespace webrtc {
     24 
     25 #define MAX_INCOMING_PAYLOAD 8096
     26 
     27 // TestPacketization callback which writes the encoded payloads to file
     28 class TestPacketization : public AudioPacketizationCallback {
     29  public:
     30   TestPacketization(RTPStream *rtpStream, uint16_t frequency);
     31   ~TestPacketization();
     32   int32_t SendData(const FrameType frameType,
     33                    const uint8_t payloadType,
     34                    const uint32_t timeStamp,
     35                    const uint8_t* payloadData,
     36                    const size_t payloadSize,
     37                    const RTPFragmentationHeader* fragmentation) override;
     38 
     39  private:
     40   static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
     41                             int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
     42   RTPStream* _rtpStream;
     43   int32_t _frequency;
     44   int16_t _seqNo;
     45 };
     46 
     47 class Sender {
     48  public:
     49   Sender();
     50   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
     51              std::string in_file_name, int sample_rate, size_t channels);
     52   void Teardown();
     53   void Run();
     54   bool Add10MsData();
     55 
     56   //for auto_test and logging
     57   uint8_t testMode;
     58   uint8_t codeId;
     59 
     60  protected:
     61   AudioCodingModule* _acm;
     62 
     63  private:
     64   PCMFile _pcmFile;
     65   AudioFrame _audioFrame;
     66   TestPacketization* _packetization;
     67 };
     68 
     69 class Receiver {
     70  public:
     71   Receiver();
     72   virtual ~Receiver() {};
     73   void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
     74              std::string out_file_name, size_t channels);
     75   void Teardown();
     76   void Run();
     77   virtual bool IncomingPacket();
     78   bool PlayoutData();
     79 
     80   //for auto_test and logging
     81   uint8_t codeId;
     82   uint8_t testMode;
     83 
     84  private:
     85   PCMFile _pcmFile;
     86   int16_t* _playoutBuffer;
     87   uint16_t _playoutLengthSmpls;
     88   int32_t _frequency;
     89   bool _firstTime;
     90 
     91  protected:
     92   AudioCodingModule* _acm;
     93   uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
     94   RTPStream* _rtpStream;
     95   WebRtcRTPHeader _rtpInfo;
     96   size_t _realPayloadSizeBytes;
     97   size_t _payloadSizeBytes;
     98   uint32_t _nextTime;
     99 };
    100 
    101 class EncodeDecodeTest : public ACMTest {
    102  public:
    103   EncodeDecodeTest();
    104   explicit EncodeDecodeTest(int testMode);
    105   void Perform() override;
    106 
    107   uint16_t _playoutFreq;
    108   uint8_t _testMode;
    109 
    110  private:
    111   std::string EncodeToFile(int fileType,
    112                            int codeId,
    113                            int* codePars,
    114                            int testMode);
    115 
    116  protected:
    117   Sender _sender;
    118   Receiver _receiver;
    119 };
    120 
    121 }  // namespace webrtc
    122 
    123 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
    124