src/third_party/webrtc/modules/audio_coding/neteq4/tools/neteq_quality_test.cc

   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 #include <stdio.h>
  12 #include "webrtc/modules/audio_coding/neteq4/tools/neteq_quality_test.h"
  13
  14 namespace webrtc {
  15 namespace test {
  16
  17 const uint8_t kPayloadType = 95;
  18 const int kOutputSizeMs = 10;
  19
  20 NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
  21                                    int in_sampling_khz,
  22                                    int out_sampling_khz,
  23                                    enum NetEqDecoder decoder_type,
  24                                    int channels,
  25                                    double drift_factor,
  26                                    std::string in_filename,
  27                                    std::string out_filename)
  28     : decoded_time_ms_(0),
  29       decodable_time_ms_(0),
  30       drift_factor_(drift_factor),
  31       block_duration_ms_(block_duration_ms),
  32       in_sampling_khz_(in_sampling_khz),
  33       out_sampling_khz_(out_sampling_khz),
  34       decoder_type_(decoder_type),
  35       channels_(channels),
  36       in_filename_(in_filename),
  37       out_filename_(out_filename),
  38       in_size_samples_(in_sampling_khz_ * block_duration_ms_),
  39       out_size_samples_(out_sampling_khz_ * kOutputSizeMs),
  40       payload_size_bytes_(0),
  41       max_payload_bytes_(0),
  42       in_file_(new InputAudioFile(in_filename_)),
  43       out_file_(NULL),
  44       rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0,
  45                                       decodable_time_ms_)),
  46       neteq_(NetEq::Create(out_sampling_khz_ * 1000)) {
  47   max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
  48   in_data_.reset(new int16_t[in_size_samples_ * channels_]);
  49   payload_.reset(new uint8_t[max_payload_bytes_]);
  50   out_data_.reset(new int16_t[out_size_samples_ * channels_]);
  51 }
  52
  53 void NetEqQualityTest::SetUp() {
  54   out_file_ = fopen(out_filename_.c_str(), "wb");
  55   ASSERT_TRUE(out_file_ != NULL);
  56   ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType));
  57   rtp_generator_->set_drift_factor(drift_factor_);
  58 }
  59
  60 void NetEqQualityTest::TearDown() {
  61   fclose(out_file_);
  62 }
  63
  64 int NetEqQualityTest::Transmit() {
  65   int packet_input_time_ms =
  66       rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
  67                                    &rtp_header_);
  68   if (!PacketLost(packet_input_time_ms) && payload_size_bytes_ > 0) {
  69     int ret = neteq_->InsertPacket(rtp_header_, &payload_[0],
  70                                    payload_size_bytes_,
  71                                    packet_input_time_ms * in_sampling_khz_);
  72     if (ret != NetEq::kOK)
  73       return -1;
  74   }
  75   return packet_input_time_ms;
  76 }
  77
  78 int NetEqQualityTest::DecodeBlock() {
  79   int channels;
  80   int samples;
  81   int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
  82                              &samples, &channels, NULL);
  83
  84   if (ret != NetEq::kOK) {
  85     return -1;
  86   } else {
  87     assert(channels == channels_);
  88     assert(samples == kOutputSizeMs * out_sampling_khz_);
  89     fwrite(&out_data_[0], sizeof(int16_t), samples * channels, out_file_);
  90     return samples;
  91   }
  92 }
  93
  94 void NetEqQualityTest::Simulate(int end_time_ms) {
  95   int audio_size_samples;
  96
  97   while (decoded_time_ms_ < end_time_ms) {
  98     while (decodable_time_ms_ - kOutputSizeMs < decoded_time_ms_) {
  99       ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
 100       payload_size_bytes_ = EncodeBlock(&in_data_[0],
 101                                         in_size_samples_, &payload_[0],
 102                                         max_payload_bytes_);
 103       decodable_time_ms_ = Transmit() + block_duration_ms_;
 104     }
 105     audio_size_samples = DecodeBlock();
 106     if (audio_size_samples > 0) {
 107       decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
 108     }
 109   }
 110 }
 111
 112 }  // namespace test
 113 }  // namespace webrtc