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