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 <math.h> 12#include <stdio.h> 13#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h" 14 15namespace webrtc { 16namespace test { 17 18const uint8_t kPayloadType = 95; 19const int kOutputSizeMs = 10; 20const int kInitSeed = 0x12345678; 21const int kPacketLossTimeUnitMs = 10; 22 23// Define switch for packet loss rate. 24static bool ValidatePacketLossRate(const char* /* flag_name */, int32_t value) { 25 if (value >= 0 && value <= 100) 26 return true; 27 printf("Invalid packet loss percentile, should be between 0 and 100."); 28 return false; 29} 30 31DEFINE_int32(packet_loss_rate, 10, "Percentile of packet loss."); 32 33static const bool packet_loss_rate_dummy = 34 RegisterFlagValidator(&FLAGS_packet_loss_rate, &ValidatePacketLossRate); 35 36// Define switch for random loss mode. 37static bool ValidateRandomLossMode(const char* /* flag_name */, int32_t value) { 38 if (value >= 0 && value <= 2) 39 return true; 40 printf("Invalid random packet loss mode, should be between 0 and 2."); 41 return false; 42} 43 44DEFINE_int32(random_loss_mode, 1, 45 "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot loss."); 46static const bool random_loss_mode_dummy = 47 RegisterFlagValidator(&FLAGS_random_loss_mode, &ValidateRandomLossMode); 48 49// Define switch for burst length. 50static bool ValidateBurstLength(const char* /* flag_name */, int32_t value) { 51 if (value >= kPacketLossTimeUnitMs) 52 return true; 53 printf("Invalid burst length, should be greater than %d ms.", 54 kPacketLossTimeUnitMs); 55 return false; 56} 57 58DEFINE_int32(burst_length, 30, 59 "Burst length in milliseconds, only valid for Gilbert Elliot loss."); 60 61static const bool burst_length_dummy = 62 RegisterFlagValidator(&FLAGS_burst_length, &ValidateBurstLength); 63 64// Define switch for drift factor. 65static bool ValidateDriftFactor(const char* /* flag_name */, double value) { 66 if (value > -0.1) 67 return true; 68 printf("Invalid drift factor, should be greater than -0.1."); 69 return false; 70} 71 72DEFINE_double(drift_factor, 0.0, "Time drift factor."); 73 74static const bool drift_factor_dummy = 75 RegisterFlagValidator(&FLAGS_drift_factor, &ValidateDriftFactor); 76 77// ProbTrans00Solver() is to calculate the transition probability from no-loss 78// state to itself in a modified Gilbert Elliot packet loss model. The result is 79// to achieve the target packet loss rate |loss_rate|, when a packet is not 80// lost only if all |units| drawings within the duration of the packet result in 81// no-loss. 82static double ProbTrans00Solver(int units, double loss_rate, 83 double prob_trans_10) { 84 if (units == 1) 85 return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10; 86// 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 * 87// prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10). 88// There is a unique solution between 0.0 and 1.0, due to the monotonicity and 89// an opposite sign at 0.0 and 1.0. 90// For simplicity, we reformulate the equation as 91// f(x) = x ^ (units - 1) + a x + b. 92// Its derivative is 93// f'(x) = (units - 1) x ^ (units - 2) + a. 94// The derivative is strictly greater than 0 when x is between 0 and 1. 95// We use Newton's method to solve the equation, iteration is 96// x(k+1) = x(k) - f(x) / f'(x); 97 const double kPrecision = 0.001f; 98 const int kIterations = 100; 99 const double a = (1.0f - loss_rate) / prob_trans_10; 100 const double b = (loss_rate - 1.0f) * (1.0f + 1.0f / prob_trans_10); 101 double x = 0.0f; // Starting point; 102 double f = b; 103 double f_p; 104 int iter = 0; 105 while ((f >= kPrecision || f <= -kPrecision) && iter < kIterations) { 106 f_p = (units - 1.0f) * pow(x, units - 2) + a; 107 x -= f / f_p; 108 if (x > 1.0f) { 109 x = 1.0f; 110 } else if (x < 0.0f) { 111 x = 0.0f; 112 } 113 f = pow(x, units - 1) + a * x + b; 114 iter ++; 115 } 116 return x; 117} 118 119NetEqQualityTest::NetEqQualityTest(int block_duration_ms, 120 int in_sampling_khz, 121 int out_sampling_khz, 122 enum NetEqDecoder decoder_type, 123 int channels, 124 std::string in_filename, 125 std::string out_filename) 126 : decoded_time_ms_(0), 127 decodable_time_ms_(0), 128 drift_factor_(FLAGS_drift_factor), 129 packet_loss_rate_(FLAGS_packet_loss_rate), 130 block_duration_ms_(block_duration_ms), 131 in_sampling_khz_(in_sampling_khz), 132 out_sampling_khz_(out_sampling_khz), 133 decoder_type_(decoder_type), 134 channels_(channels), 135 in_filename_(in_filename), 136 out_filename_(out_filename), 137 log_filename_(out_filename + ".log"), 138 in_size_samples_(in_sampling_khz_ * block_duration_ms_), 139 out_size_samples_(out_sampling_khz_ * kOutputSizeMs), 140 payload_size_bytes_(0), 141 max_payload_bytes_(0), 142 in_file_(new InputAudioFile(in_filename_)), 143 out_file_(NULL), 144 log_file_(NULL), 145 rtp_generator_(new RtpGenerator(in_sampling_khz_, 0, 0, 146 decodable_time_ms_)), 147 total_payload_size_bytes_(0) { 148 NetEq::Config config; 149 config.sample_rate_hz = out_sampling_khz_ * 1000; 150 neteq_.reset(NetEq::Create(config)); 151 max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t); 152 in_data_.reset(new int16_t[in_size_samples_ * channels_]); 153 payload_.reset(new uint8_t[max_payload_bytes_]); 154 out_data_.reset(new int16_t[out_size_samples_ * channels_]); 155} 156 157bool NoLoss::Lost() { 158 return false; 159} 160 161UniformLoss::UniformLoss(double loss_rate) 162 : loss_rate_(loss_rate) { 163} 164 165bool UniformLoss::Lost() { 166 int drop_this = rand(); 167 return (drop_this < loss_rate_ * RAND_MAX); 168} 169 170GilbertElliotLoss::GilbertElliotLoss(double prob_trans_11, double prob_trans_01) 171 : prob_trans_11_(prob_trans_11), 172 prob_trans_01_(prob_trans_01), 173 lost_last_(false), 174 uniform_loss_model_(new UniformLoss(0)) { 175} 176 177bool GilbertElliotLoss::Lost() { 178 // Simulate bursty channel (Gilbert model). 179 // (1st order) Markov chain model with memory of the previous/last 180 // packet state (lost or received). 181 if (lost_last_) { 182 // Previous packet was not received. 183 uniform_loss_model_->set_loss_rate(prob_trans_11_); 184 return lost_last_ = uniform_loss_model_->Lost(); 185 } else { 186 uniform_loss_model_->set_loss_rate(prob_trans_01_); 187 return lost_last_ = uniform_loss_model_->Lost(); 188 } 189} 190 191void NetEqQualityTest::SetUp() { 192 out_file_ = fopen(out_filename_.c_str(), "wb"); 193 log_file_ = fopen(log_filename_.c_str(), "wt"); 194 ASSERT_TRUE(out_file_ != NULL); 195 ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType)); 196 rtp_generator_->set_drift_factor(drift_factor_); 197 198 int units = block_duration_ms_ / kPacketLossTimeUnitMs; 199 switch (FLAGS_random_loss_mode) { 200 case 1: { 201 // |unit_loss_rate| is the packet loss rate for each unit time interval 202 // (kPacketLossTimeUnitMs). Since a packet loss event is generated if any 203 // of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of 204 // a full packet duration is drawn with a loss, |unit_loss_rate| fulfills 205 // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) == 206 // 1 - packet_loss_rate. 207 double unit_loss_rate = (1.0f - pow(1.0f - 0.01f * packet_loss_rate_, 208 1.0f / units)); 209 loss_model_.reset(new UniformLoss(unit_loss_rate)); 210 break; 211 } 212 case 2: { 213 // |FLAGS_burst_length| should be integer times of kPacketLossTimeUnitMs. 214 ASSERT_EQ(0, FLAGS_burst_length % kPacketLossTimeUnitMs); 215 216 // We do not allow 100 percent packet loss in Gilbert Elliot model, which 217 // makes no sense. 218 ASSERT_GT(100, packet_loss_rate_); 219 220 // To guarantee the overall packet loss rate, transition probabilities 221 // need to satisfy: 222 // pi_0 * (1 - prob_trans_01_) ^ units + 223 // pi_1 * prob_trans_10_ ^ (units - 1) == 1 - loss_rate 224 // pi_0 = prob_trans_10 / (prob_trans_10 + prob_trans_01_) 225 // is the stationary state probability of no-loss 226 // pi_1 = prob_trans_01_ / (prob_trans_10 + prob_trans_01_) 227 // is the stationary state probability of loss 228 // After a derivation prob_trans_00 should satisfy: 229 // prob_trans_00 ^ (units - 1) = (loss_rate - 1) / prob_trans_10 * 230 // prob_trans_00 + (1 - loss_rate) * (1 + 1 / prob_trans_10). 231 double loss_rate = 0.01f * packet_loss_rate_; 232 double prob_trans_10 = 1.0f * kPacketLossTimeUnitMs / FLAGS_burst_length; 233 double prob_trans_00 = ProbTrans00Solver(units, loss_rate, prob_trans_10); 234 loss_model_.reset(new GilbertElliotLoss(1.0f - prob_trans_10, 235 1.0f - prob_trans_00)); 236 break; 237 } 238 default: { 239 loss_model_.reset(new NoLoss); 240 break; 241 } 242 } 243 244 // Make sure that the packet loss profile is same for all derived tests. 245 srand(kInitSeed); 246} 247 248void NetEqQualityTest::TearDown() { 249 fclose(out_file_); 250} 251 252bool NetEqQualityTest::PacketLost() { 253 int cycles = block_duration_ms_ / kPacketLossTimeUnitMs; 254 255 // The loop is to make sure that codecs with different block lengths share the 256 // same packet loss profile. 257 bool lost = false; 258 for (int idx = 0; idx < cycles; idx ++) { 259 if (loss_model_->Lost()) { 260 // The packet will be lost if any of the drawings indicates a loss, but 261 // the loop has to go on to make sure that codecs with different block 262 // lengths keep the same pace. 263 lost = true; 264 } 265 } 266 return lost; 267} 268 269int NetEqQualityTest::Transmit() { 270 int packet_input_time_ms = 271 rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_, 272 &rtp_header_); 273 if (payload_size_bytes_ > 0) { 274 fprintf(log_file_, "Packet at %d ms", packet_input_time_ms); 275 if (!PacketLost()) { 276 int ret = neteq_->InsertPacket(rtp_header_, &payload_[0], 277 payload_size_bytes_, 278 packet_input_time_ms * in_sampling_khz_); 279 if (ret != NetEq::kOK) 280 return -1; 281 fprintf(log_file_, " OK.\n"); 282 } else { 283 fprintf(log_file_, " Lost.\n"); 284 } 285 } 286 return packet_input_time_ms; 287} 288 289int NetEqQualityTest::DecodeBlock() { 290 int channels; 291 int samples; 292 int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0], 293 &samples, &channels, NULL); 294 295 if (ret != NetEq::kOK) { 296 return -1; 297 } else { 298 assert(channels == channels_); 299 assert(samples == kOutputSizeMs * out_sampling_khz_); 300 fwrite(&out_data_[0], sizeof(int16_t), samples * channels, out_file_); 301 return samples; 302 } 303} 304 305void NetEqQualityTest::Simulate(int end_time_ms) { 306 int audio_size_samples; 307 308 while (decoded_time_ms_ < end_time_ms) { 309 // Assume 10 packets in packets buffer. 310 while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) { 311 ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0])); 312 payload_size_bytes_ = EncodeBlock(&in_data_[0], 313 in_size_samples_, &payload_[0], 314 max_payload_bytes_); 315 total_payload_size_bytes_ += payload_size_bytes_; 316 decodable_time_ms_ = Transmit() + block_duration_ms_; 317 } 318 audio_size_samples = DecodeBlock(); 319 if (audio_size_samples > 0) { 320 decoded_time_ms_ += audio_size_samples / out_sampling_khz_; 321 } 322 } 323 fprintf(log_file_, "%f", 8.0f * total_payload_size_bytes_ / end_time_ms); 324} 325 326} // namespace test 327} // namespace webrtc 328