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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 <string>
     15 #include "testing/gtest/include/gtest/gtest.h"
     16 #include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
     17 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
     18 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
     19 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
     20 #include "webrtc/typedefs.h"
     21 
     22 namespace webrtc {
     23 namespace test {
     24 
     25 class NetEqQualityTest : public ::testing::Test {
     26  protected:
     27   NetEqQualityTest(int block_duration_ms,
     28                    int in_sampling_khz,
     29                    int out_sampling_khz,
     30                    enum NetEqDecoder decoder_type,
     31                    int channels,
     32                    double drift_factor,
     33                    std::string in_filename,
     34                    std::string out_filename);
     35   virtual void SetUp() OVERRIDE;
     36   virtual void TearDown() OVERRIDE;
     37 
     38   // EncodeBlock(...) does the following:
     39   // 1. encodes a block of audio, saved in |in_data| and has a length of
     40   // |block_size_samples| (samples per channel),
     41   // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
     42   // 3. returns the length of the payload (in bytes),
     43   virtual int EncodeBlock(int16_t* in_data, int block_size_samples,
     44                           uint8_t* payload, int max_bytes) = 0;
     45 
     46   // PacketLoss(...) determines weather a packet sent at an indicated time gets
     47   // lost or not.
     48   virtual bool PacketLost(int packet_input_time_ms) { return false; }
     49 
     50   // DecodeBlock() decodes a block of audio using the payload stored in
     51   // |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
     52   // audio is to be stored in |out_data_|.
     53   int DecodeBlock();
     54 
     55   // Transmit() uses |rtp_generator_| to generate a packet and passes it to
     56   // |neteq_|.
     57   int Transmit();
     58 
     59   // Simulate(...) runs encoding / transmitting / decoding up to |end_time_ms|
     60   // (miliseconds), the resulted audio is stored in the file with the name of
     61   // |out_filename_|.
     62   void Simulate(int end_time_ms);
     63 
     64  private:
     65   int decoded_time_ms_;
     66   int decodable_time_ms_;
     67   double drift_factor_;
     68   const int block_duration_ms_;
     69   const int in_sampling_khz_;
     70   const int out_sampling_khz_;
     71   const enum NetEqDecoder decoder_type_;
     72   const int channels_;
     73   const std::string in_filename_;
     74   const std::string out_filename_;
     75 
     76   // Number of samples per channel in a frame.
     77   const int in_size_samples_;
     78 
     79   // Expected output number of samples per channel in a frame.
     80   const int out_size_samples_;
     81 
     82   int payload_size_bytes_;
     83   int max_payload_bytes_;
     84 
     85   scoped_ptr<InputAudioFile> in_file_;
     86   FILE* out_file_;
     87 
     88   scoped_ptr<RtpGenerator> rtp_generator_;
     89   scoped_ptr<NetEq> neteq_;
     90 
     91   scoped_ptr<int16_t[]> in_data_;
     92   scoped_ptr<uint8_t[]> payload_;
     93   scoped_ptr<int16_t[]> out_data_;
     94   WebRtcRTPHeader rtp_header_;
     95 };
     96 
     97 }  // namespace test
     98 }  // namespace webrtc
     99 
    100 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
    101