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      1 /*
      2  *  Copyright (c) 2013 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 <assert.h>
     12 #include <math.h>
     13 
     14 #include <iostream>
     15 
     16 #include "gflags/gflags.h"
     17 #include "testing/gtest/include/gtest/gtest.h"
     18 #include "webrtc/common.h"
     19 #include "webrtc/common_types.h"
     20 #include "webrtc/engine_configurations.h"
     21 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
     22 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
     23 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
     24 #include "webrtc/modules/audio_coding/main/test/Channel.h"
     25 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
     26 #include "webrtc/modules/audio_coding/main/test/utility.h"
     27 #include "webrtc/system_wrappers/interface/event_wrapper.h"
     28 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
     29 #include "webrtc/test/testsupport/fileutils.h"
     30 
     31 DEFINE_string(codec, "isac", "Codec Name");
     32 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
     33 DEFINE_int32(num_channels, 1, "Number of Channels.");
     34 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
     35 DEFINE_int32(delay, 0, "Delay in millisecond.");
     36 DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
     37 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
     38 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
     39 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
     40 
     41 namespace webrtc {
     42 
     43 namespace {
     44 
     45 struct CodecSettings {
     46   char name[50];
     47   int sample_rate_hz;
     48   int num_channels;
     49 };
     50 
     51 struct AcmSettings {
     52   bool dtx;
     53   bool fec;
     54 };
     55 
     56 struct TestSettings {
     57   CodecSettings codec;
     58   AcmSettings acm;
     59   bool packet_loss;
     60 };
     61 
     62 }  // namespace
     63 
     64 class DelayTest {
     65  public:
     66   DelayTest()
     67       : acm_a_(AudioCodingModule::Create(0)),
     68         acm_b_(AudioCodingModule::Create(1)),
     69         channel_a2b_(new Channel),
     70         test_cntr_(0),
     71         encoding_sample_rate_hz_(8000) {}
     72 
     73   ~DelayTest() {
     74     if (channel_a2b_ != NULL) {
     75       delete channel_a2b_;
     76       channel_a2b_ = NULL;
     77     }
     78     in_file_a_.Close();
     79   }
     80 
     81   void Initialize() {
     82     test_cntr_ = 0;
     83     std::string file_name = webrtc::test::ResourcePath(
     84         "audio_coding/testfile32kHz", "pcm");
     85     if (FLAGS_input_file.size() > 0)
     86       file_name = FLAGS_input_file;
     87     in_file_a_.Open(file_name, 32000, "rb");
     88     ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
     89         "Couldn't initialize receiver.\n";
     90     ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
     91         "Couldn't initialize receiver.\n";
     92     if (FLAGS_init_delay > 0) {
     93       ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
     94           "Failed to set initial delay.\n";
     95     }
     96 
     97     if (FLAGS_delay > 0) {
     98       ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
     99           "Failed to set minimum delay.\n";
    100     }
    101 
    102     int num_encoders = acm_a_->NumberOfCodecs();
    103     CodecInst my_codec_param;
    104     for (int n = 0; n < num_encoders; n++) {
    105       EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
    106           "Failed to get codec.";
    107       if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
    108         my_codec_param.channels = 1;
    109       else if (my_codec_param.channels > 1)
    110         continue;
    111       if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
    112           my_codec_param.plfreq == 48000)
    113         continue;
    114       if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
    115         continue;
    116       ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
    117           "Couldn't register receive codec.\n";
    118     }
    119 
    120     // Create and connect the channel
    121     ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
    122         "Couldn't register Transport callback.\n";
    123     channel_a2b_->RegisterReceiverACM(acm_b_.get());
    124   }
    125 
    126   void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
    127                const char* output_prefix) {
    128     for (size_t n = 0; n < num_tests; ++n) {
    129       ApplyConfig(config[n]);
    130       Run(duration_sec, output_prefix);
    131     }
    132   }
    133 
    134  private:
    135   void ApplyConfig(const TestSettings& config) {
    136     printf("====================================\n");
    137     printf("Test %d \n"
    138            "Codec: %s, %d kHz, %d channel(s)\n"
    139            "ACM: DTX %s, FEC %s\n"
    140            "Channel: %s\n",
    141            ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
    142            config.codec.num_channels, config.acm.dtx ? "on" : "off",
    143            config.acm.fec ? "on" : "off",
    144            config.packet_loss ? "with packet-loss" : "no packet-loss");
    145     SendCodec(config.codec);
    146     ConfigAcm(config.acm);
    147     ConfigChannel(config.packet_loss);
    148   }
    149 
    150   void SendCodec(const CodecSettings& config) {
    151     CodecInst my_codec_param;
    152     ASSERT_EQ(0, AudioCodingModule::Codec(
    153               config.name, &my_codec_param, config.sample_rate_hz,
    154               config.num_channels)) << "Specified codec is not supported.\n";
    155 
    156     encoding_sample_rate_hz_ = my_codec_param.plfreq;
    157     ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
    158         "Failed to register send-codec.\n";
    159   }
    160 
    161   void ConfigAcm(const AcmSettings& config) {
    162     ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
    163         "Failed to set VAD.\n";
    164     ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
    165         "Failed to set RED.\n";
    166   }
    167 
    168   void ConfigChannel(bool packet_loss) {
    169     channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
    170   }
    171 
    172   void OpenOutFile(const char* output_id) {
    173     std::stringstream file_stream;
    174     file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
    175         << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
    176     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
    177     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
    178     out_file_b_.Open(file_name.c_str(), 32000, "wb");
    179   }
    180 
    181   void Run(int duration_sec, const char* output_prefix) {
    182     OpenOutFile(output_prefix);
    183     AudioFrame audio_frame;
    184     uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
    185 
    186     int num_frames = 0;
    187     int in_file_frames = 0;
    188     uint32_t playout_ts;
    189     uint32_t received_ts;
    190     double average_delay = 0;
    191     double inst_delay_sec = 0;
    192     while (num_frames < (duration_sec * 100)) {
    193       if (in_file_a_.EndOfFile()) {
    194         in_file_a_.Rewind();
    195       }
    196 
    197       // Print delay information every 16 frame
    198       if ((num_frames & 0x3F) == 0x3F) {
    199         ACMNetworkStatistics statistics;
    200         acm_b_->NetworkStatistics(&statistics);
    201         fprintf(stdout, "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
    202                 " ts-based average = %6.3f, "
    203                 "curr buff-lev = %4u opt buff-lev = %4u \n",
    204                 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
    205                 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
    206                 average_delay, statistics.currentBufferSize,
    207                 statistics.preferredBufferSize);
    208         fflush (stdout);
    209       }
    210 
    211       in_file_a_.Read10MsData(audio_frame);
    212       ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
    213       ASSERT_LE(0, acm_a_->Process());
    214       ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
    215       out_file_b_.Write10MsData(
    216           audio_frame.data_,
    217           audio_frame.samples_per_channel_ * audio_frame.num_channels_);
    218       acm_b_->PlayoutTimestamp(&playout_ts);
    219       received_ts = channel_a2b_->LastInTimestamp();
    220       inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
    221           / static_cast<double>(encoding_sample_rate_hz_);
    222 
    223       if (num_frames > 10)
    224         average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
    225 
    226       ++num_frames;
    227       ++in_file_frames;
    228     }
    229     out_file_b_.Close();
    230   }
    231 
    232   scoped_ptr<AudioCodingModule> acm_a_;
    233   scoped_ptr<AudioCodingModule> acm_b_;
    234 
    235   Channel* channel_a2b_;
    236 
    237   PCMFile in_file_a_;
    238   PCMFile out_file_b_;
    239   int test_cntr_;
    240   int encoding_sample_rate_hz_;
    241 };
    242 
    243 }  // namespace webrtc
    244 
    245 int main(int argc, char* argv[]) {
    246   google::ParseCommandLineFlags(&argc, &argv, true);
    247   webrtc::TestSettings test_setting;
    248   strcpy(test_setting.codec.name, FLAGS_codec.c_str());
    249 
    250   if (FLAGS_sample_rate_hz != 8000 &&
    251       FLAGS_sample_rate_hz != 16000 &&
    252       FLAGS_sample_rate_hz != 32000 &&
    253       FLAGS_sample_rate_hz != 48000) {
    254     std::cout << "Invalid sampling rate.\n";
    255     return 1;
    256   }
    257   test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
    258   if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
    259     std::cout << "Only mono and stereo are supported.\n";
    260     return 1;
    261   }
    262   test_setting.codec.num_channels = FLAGS_num_channels;
    263   test_setting.acm.dtx = FLAGS_dtx;
    264   test_setting.acm.fec = FLAGS_fec;
    265   test_setting.packet_loss = FLAGS_packet_loss;
    266 
    267   webrtc::DelayTest delay_test;
    268   delay_test.Initialize();
    269   delay_test.Perform(&test_setting, 1, 240, "delay_test");
    270   return 0;
    271 }
    272