audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/test/delay_test.cc

Index: webrtc/modules/audio_coding/main/test/delay_test.cc
diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/main/test/delay_test.cc
deleted file mode 100644
index ce08c0f4a2170e247de04236e7d5d7d0704e82a0..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/delay_test.cc
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <math.h>
-
-#include <iostream>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common.h"
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-DEFINE_string(codec, "isac", "Codec Name");
-DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
-DEFINE_int32(num_channels, 1, "Number of Channels.");
-DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
-DEFINE_int32(delay, 0, "Delay in millisecond.");
-DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
-DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
-DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
-
-namespace webrtc {
-
-namespace {
-
-struct CodecSettings {
-  char name[50];
-  int sample_rate_hz;
-  int num_channels;
-};
-
-struct AcmSettings {
-  bool dtx;
-  bool fec;
-};
-
-struct TestSettings {
-  CodecSettings codec;
-  AcmSettings acm;
-  bool packet_loss;
-};
-
-}  // namespace
-
-class DelayTest {
- public:
-  DelayTest()
-      : acm_a_(AudioCodingModule::Create(0)),
-        acm_b_(AudioCodingModule::Create(1)),
-        channel_a2b_(new Channel),
-        test_cntr_(0),
-        encoding_sample_rate_hz_(8000) {}
-
-  ~DelayTest() {
-    if (channel_a2b_ != NULL) {
-      delete channel_a2b_;
-      channel_a2b_ = NULL;
-    }
-    in_file_a_.Close();
-  }
-
-  void Initialize() {
-    test_cntr_ = 0;
-    std::string file_name = webrtc::test::ResourcePath(
-        "audio_coding/testfile32kHz", "pcm");
-    if (FLAGS_input_file.size() > 0)
-      file_name = FLAGS_input_file;
-    in_file_a_.Open(file_name, 32000, "rb");
-    ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
-        "Couldn't initialize receiver.\n";
-    ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
-        "Couldn't initialize receiver.\n";
-
-    if (FLAGS_delay > 0) {
-      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
-          "Failed to set minimum delay.\n";
-    }
-
-    int num_encoders = acm_a_->NumberOfCodecs();
-    CodecInst my_codec_param;
-    for (int n = 0; n < num_encoders; n++) {
-      EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
-          "Failed to get codec.";
-      if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
-        my_codec_param.channels = 1;
-      else if (my_codec_param.channels > 1)
-        continue;
-      if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
-          my_codec_param.plfreq == 48000)
-        continue;
-      if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
-        continue;
-      ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
-          "Couldn't register receive codec.\n";
-    }
-
-    // Create and connect the channel
-    ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
-        "Couldn't register Transport callback.\n";
-    channel_a2b_->RegisterReceiverACM(acm_b_.get());
-  }
-
-  void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
-               const char* output_prefix) {
-    for (size_t n = 0; n < num_tests; ++n) {
-      ApplyConfig(config[n]);
-      Run(duration_sec, output_prefix);
-    }
-  }
-
- private:
-  void ApplyConfig(const TestSettings& config) {
-    printf("====================================\n");
-    printf("Test %d \n"
-           "Codec: %s, %d kHz, %d channel(s)\n"
-           "ACM: DTX %s, FEC %s\n"
-           "Channel: %s\n",
-           ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
-           config.codec.num_channels, config.acm.dtx ? "on" : "off",
-           config.acm.fec ? "on" : "off",
-           config.packet_loss ? "with packet-loss" : "no packet-loss");
-    SendCodec(config.codec);
-    ConfigAcm(config.acm);
-    ConfigChannel(config.packet_loss);
-  }
-
-  void SendCodec(const CodecSettings& config) {
-    CodecInst my_codec_param;
-    ASSERT_EQ(0, AudioCodingModule::Codec(
-              config.name, &my_codec_param, config.sample_rate_hz,
-              config.num_channels)) << "Specified codec is not supported.\n";
-
-    encoding_sample_rate_hz_ = my_codec_param.plfreq;
-    ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
-        "Failed to register send-codec.\n";
-  }
-
-  void ConfigAcm(const AcmSettings& config) {
-    ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
-        "Failed to set VAD.\n";
-    ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
-        "Failed to set RED.\n";
-  }
-
-  void ConfigChannel(bool packet_loss) {
-    channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
-  }
-
-  void OpenOutFile(const char* output_id) {
-    std::stringstream file_stream;
-    file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
-        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
-    std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
-    std::string file_name = webrtc::test::OutputPath() + file_stream.str();
-    out_file_b_.Open(file_name.c_str(), 32000, "wb");
-  }
-
-  void Run(int duration_sec, const char* output_prefix) {
-    OpenOutFile(output_prefix);
-    AudioFrame audio_frame;
-    uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
-
-    int num_frames = 0;
-    int in_file_frames = 0;
-    uint32_t playout_ts;
-    uint32_t received_ts;
-    double average_delay = 0;
-    double inst_delay_sec = 0;
-    while (num_frames < (duration_sec * 100)) {
-      if (in_file_a_.EndOfFile()) {
-        in_file_a_.Rewind();
-      }
-
-      // Print delay information every 16 frame
-      if ((num_frames & 0x3F) == 0x3F) {
-        NetworkStatistics statistics;
-        acm_b_->GetNetworkStatistics(&statistics);
-        fprintf(stdout, "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
-                " ts-based average = %6.3f, "
-                "curr buff-lev = %4u opt buff-lev = %4u \n",
-                statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
-                statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
-                average_delay, statistics.currentBufferSize,
-                statistics.preferredBufferSize);
-        fflush (stdout);
-      }
-
-      in_file_a_.Read10MsData(audio_frame);
-      ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
-      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
-      out_file_b_.Write10MsData(
-          audio_frame.data_,
-          audio_frame.samples_per_channel_ * audio_frame.num_channels_);
-      acm_b_->PlayoutTimestamp(&playout_ts);
-      received_ts = channel_a2b_->LastInTimestamp();
-      inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
-          / static_cast<double>(encoding_sample_rate_hz_);
-
-      if (num_frames > 10)
-        average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
-
-      ++num_frames;
-      ++in_file_frames;
-    }
-    out_file_b_.Close();
-  }
-
-  rtc::scoped_ptr<AudioCodingModule> acm_a_;
-  rtc::scoped_ptr<AudioCodingModule> acm_b_;
-
-  Channel* channel_a2b_;
-
-  PCMFile in_file_a_;
-  PCMFile out_file_b_;
-  int test_cntr_;
-  int encoding_sample_rate_hz_;
-};
-
-}  // namespace webrtc
-
-int main(int argc, char* argv[]) {
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::TestSettings test_setting;
-  strcpy(test_setting.codec.name, FLAGS_codec.c_str());
-
-  if (FLAGS_sample_rate_hz != 8000 &&
-      FLAGS_sample_rate_hz != 16000 &&
-      FLAGS_sample_rate_hz != 32000 &&
-      FLAGS_sample_rate_hz != 48000) {
-    std::cout << "Invalid sampling rate.\n";
-    return 1;
-  }
-  test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
-  if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
-    std::cout << "Only mono and stereo are supported.\n";
-    return 1;
-  }
-  test_setting.codec.num_channels = FLAGS_num_channels;
-  test_setting.acm.dtx = FLAGS_dtx;
-  test_setting.acm.fec = FLAGS_fec;
-  test_setting.packet_loss = FLAGS_packet_loss;
-
-  webrtc::DelayTest delay_test;
-  delay_test.Initialize();
-  delay_test.Perform(&test_setting, 1, 240, "delay_test");
-  return 0;
-}