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      1 /*
      2  *  Copyright (c) 2014 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_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
     13 
     14 #include <fstream>
     15 #include <gflags/gflags.h>
     16 #include "testing/gtest/include/gtest/gtest.h"
     17 #include "webrtc/base/scoped_ptr.h"
     18 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
     19 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
     20 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
     21 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
     22 #include "webrtc/typedefs.h"
     23 
     24 using google::RegisterFlagValidator;
     25 
     26 namespace webrtc {
     27 namespace test {
     28 
     29 class LossModel {
     30  public:
     31   virtual ~LossModel() {};
     32   virtual bool Lost() = 0;
     33 };
     34 
     35 class NoLoss : public LossModel {
     36  public:
     37   bool Lost() override;
     38 };
     39 
     40 class UniformLoss : public LossModel {
     41  public:
     42   UniformLoss(double loss_rate);
     43   bool Lost() override;
     44   void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
     45 
     46  private:
     47   double loss_rate_;
     48 };
     49 
     50 class GilbertElliotLoss : public LossModel {
     51  public:
     52   GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
     53   bool Lost() override;
     54 
     55  private:
     56   // Prob. of losing current packet, when previous packet is lost.
     57   double prob_trans_11_;
     58   // Prob. of losing current packet, when previous packet is not lost.
     59   double prob_trans_01_;
     60   bool lost_last_;
     61   rtc::scoped_ptr<UniformLoss> uniform_loss_model_;
     62 };
     63 
     64 class NetEqQualityTest : public ::testing::Test {
     65  protected:
     66   NetEqQualityTest(int block_duration_ms,
     67                    int in_sampling_khz,
     68                    int out_sampling_khz,
     69                    NetEqDecoder decoder_type);
     70   virtual ~NetEqQualityTest();
     71 
     72   void SetUp() override;
     73 
     74   // EncodeBlock(...) does the following:
     75   // 1. encodes a block of audio, saved in |in_data| and has a length of
     76   // |block_size_samples| (samples per channel),
     77   // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
     78   // 3. returns the length of the payload (in bytes),
     79   virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
     80                           uint8_t* payload, size_t max_bytes) = 0;
     81 
     82   // PacketLost(...) determines weather a packet sent at an indicated time gets
     83   // lost or not.
     84   bool PacketLost();
     85 
     86   // DecodeBlock() decodes a block of audio using the payload stored in
     87   // |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
     88   // audio is to be stored in |out_data_|.
     89   int DecodeBlock();
     90 
     91   // Transmit() uses |rtp_generator_| to generate a packet and passes it to
     92   // |neteq_|.
     93   int Transmit();
     94 
     95   // Runs encoding / transmitting / decoding.
     96   void Simulate();
     97 
     98   // Write to log file. Usage Log() << ...
     99   std::ofstream& Log();
    100 
    101   NetEqDecoder decoder_type_;
    102   const size_t channels_;
    103 
    104  private:
    105   int decoded_time_ms_;
    106   int decodable_time_ms_;
    107   double drift_factor_;
    108   int packet_loss_rate_;
    109   const int block_duration_ms_;
    110   const int in_sampling_khz_;
    111   const int out_sampling_khz_;
    112 
    113   // Number of samples per channel in a frame.
    114   const size_t in_size_samples_;
    115 
    116   // Expected output number of samples per channel in a frame.
    117   const size_t out_size_samples_;
    118 
    119   size_t payload_size_bytes_;
    120   size_t max_payload_bytes_;
    121 
    122   rtc::scoped_ptr<InputAudioFile> in_file_;
    123   rtc::scoped_ptr<AudioSink> output_;
    124   std::ofstream log_file_;
    125 
    126   rtc::scoped_ptr<RtpGenerator> rtp_generator_;
    127   rtc::scoped_ptr<NetEq> neteq_;
    128   rtc::scoped_ptr<LossModel> loss_model_;
    129 
    130   rtc::scoped_ptr<int16_t[]> in_data_;
    131   rtc::scoped_ptr<uint8_t[]> payload_;
    132   rtc::scoped_ptr<int16_t[]> out_data_;
    133   WebRtcRTPHeader rtp_header_;
    134 
    135   size_t total_payload_size_bytes_;
    136 };
    137 
    138 }  // namespace test
    139 }  // namespace webrtc
    140 
    141 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
    142