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
31DEFINE_string(codec, "isac", "Codec Name");
32DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
33DEFINE_int32(num_channels, 1, "Number of Channels.");
34DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
35DEFINE_int32(delay, 0, "Delay in millisecond.");
36DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
37DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
38DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
39
40namespace webrtc {
41
42namespace {
43
44struct CodecSettings {
45  char name[50];
46  int sample_rate_hz;
47  int num_channels;
48};
49
50struct AcmSettings {
51  bool dtx;
52  bool fec;
53};
54
55struct TestSettings {
56  CodecSettings codec;
57  AcmSettings acm;
58  bool packet_loss;
59};
60
61}  // namespace
62
63class 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
239int 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