audio_coding: remove "main" directory

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

Index: webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
deleted file mode 100644
index 857381d250d011ebe78a0f8d350bdae894c3833b..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ /dev/null
@@ -1,307 +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 <stdio.h>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-// Codec.
-DEFINE_string(codec, "opus", "Codec Name");
-DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
-DEFINE_int32(codec_channels, 1, "Number of channels of the codec.");
-
-// PCM input/output.
-DEFINE_string(input, "", "Input PCM file at 16 kHz.");
-DEFINE_bool(input_stereo, false, "Input is stereo.");
-DEFINE_int32(input_fs_hz, 32000, "Input sample rate Hz.");
-DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
-DEFINE_int32(output_fs_hz, 32000, "Output sample rate Hz");
-
-// Timing files
-DEFINE_string(seq_num, "seq_num", "Sequence number file.");
-DEFINE_string(send_ts, "send_timestamp", "Send timestamp file.");
-DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
-
-// Delay logging
-DEFINE_string(delay, "", "Log for delay.");
-
-// Other setups
-DEFINE_bool(verbose, false, "Verbosity.");
-DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
-
-const int32_t kAudioPlayedOut = 0x00000001;
-const int32_t kPacketPushedIn = 0x00000001 << 1;
-const int kPlayoutPeriodMs = 10;
-
-namespace webrtc {
-
-class InsertPacketWithTiming {
- public:
-  InsertPacketWithTiming()
-      : sender_clock_(new SimulatedClock(0)),
-        receiver_clock_(new SimulatedClock(0)),
-        send_acm_(AudioCodingModule::Create(0, sender_clock_)),
-        receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
-        channel_(new Channel),
-        seq_num_fid_(fopen(FLAGS_seq_num.c_str(), "rt")),
-        send_ts_fid_(fopen(FLAGS_send_ts.c_str(), "rt")),
-        receive_ts_fid_(fopen(FLAGS_receive_ts.c_str(), "rt")),
-        pcm_out_fid_(fopen(FLAGS_output.c_str(), "wb")),
-        samples_in_1ms_(48),
-        num_10ms_in_codec_frame_(2),  // Typical 20 ms frames.
-        time_to_insert_packet_ms_(3),  // An arbitrary offset on pushing packet.
-        next_receive_ts_(0),
-        time_to_playout_audio_ms_(kPlayoutPeriodMs),
-        loss_threshold_(0),
-        playout_timing_fid_(fopen("playout_timing.txt", "wt")) {}
-
-  void SetUp() {
-    ASSERT_TRUE(sender_clock_ != NULL);
-    ASSERT_TRUE(receiver_clock_ != NULL);
-
-    ASSERT_TRUE(send_acm_.get() != NULL);
-    ASSERT_TRUE(receive_acm_.get() != NULL);
-    ASSERT_TRUE(channel_ != NULL);
-
-    ASSERT_TRUE(seq_num_fid_ != NULL);
-    ASSERT_TRUE(send_ts_fid_ != NULL);
-    ASSERT_TRUE(receive_ts_fid_ != NULL);
-
-    ASSERT_TRUE(playout_timing_fid_ != NULL);
-
-    next_receive_ts_ = ReceiveTimestamp();
-
-    CodecInst codec;
-    ASSERT_EQ(0, AudioCodingModule::Codec(FLAGS_codec.c_str(), &codec,
-                             FLAGS_codec_sample_rate_hz,
-                             FLAGS_codec_channels));
-    ASSERT_EQ(0, receive_acm_->InitializeReceiver());
-    ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
-    ASSERT_EQ(0, receive_acm_->RegisterReceiveCodec(codec));
-
-    // Set codec-dependent parameters.
-    samples_in_1ms_ = codec.plfreq / 1000;
-    num_10ms_in_codec_frame_ = codec.pacsize / (codec.plfreq / 100);
-
-    channel_->RegisterReceiverACM(receive_acm_.get());
-    send_acm_->RegisterTransportCallback(channel_);
-
-    if (FLAGS_input.size() == 0) {
-      std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
-                                                 "pcm");
-      pcm_in_fid_.Open(file_name, 32000, "r", true);  // auto-rewind
-      std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
-    } else {
-      pcm_in_fid_.Open(FLAGS_input, static_cast<uint16_t>(FLAGS_input_fs_hz),
-                    "r", true);  // auto-rewind
-      std::cout << "Input file " << FLAGS_input << "at " << FLAGS_input_fs_hz
-          << " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.")
-          << std::endl;
-      pcm_in_fid_.ReadStereo(FLAGS_input_stereo);
-    }
-
-    ASSERT_TRUE(pcm_out_fid_ != NULL);
-    std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz
-        << " Hz." << std::endl;
-
-    // Other setups
-    if (FLAGS_loss_rate > 0)
-      loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
-    else
-      loss_threshold_ = 0;
-  }
-
-  void TickOneMillisecond(uint32_t* action) {
-    // One millisecond passed.
-    time_to_insert_packet_ms_--;
-    time_to_playout_audio_ms_--;
-    sender_clock_->AdvanceTimeMilliseconds(1);
-    receiver_clock_->AdvanceTimeMilliseconds(1);
-
-    // Reset action.
-    *action = 0;
-
-    // Is it time to pull audio?
-    if (time_to_playout_audio_ms_ == 0) {
-      time_to_playout_audio_ms_ = kPlayoutPeriodMs;
-      receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
-                                    &frame_);
-      fwrite(frame_.data_, sizeof(frame_.data_[0]),
-             frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
-      *action |= kAudioPlayedOut;
-    }
-
-    // Is it time to push in next packet?
-    if (time_to_insert_packet_ms_ <= .5) {
-      *action |= kPacketPushedIn;
-
-      // Update time-to-insert packet.
-      uint32_t t = next_receive_ts_;
-      next_receive_ts_ = ReceiveTimestamp();
-      time_to_insert_packet_ms_ += static_cast<float>(next_receive_ts_ - t) /
-          samples_in_1ms_;
-
-      // Push in just enough audio.
-      for (int n = 0; n < num_10ms_in_codec_frame_; n++) {
-        pcm_in_fid_.Read10MsData(frame_);
-        EXPECT_GE(send_acm_->Add10MsData(frame_), 0);
-      }
-
-      // Set the parameters for the packet to be pushed in receiver ACM right
-      // now.
-      uint32_t ts = SendTimestamp();
-      int seq_num = SequenceNumber();
-      bool lost = false;
-      channel_->set_send_timestamp(ts);
-      channel_->set_sequence_number(seq_num);
-      if (loss_threshold_ > 0 && rand() < loss_threshold_) {
-        channel_->set_num_packets_to_drop(1);
-        lost = true;
-      }
-
-      if (FLAGS_verbose) {
-        if (!lost) {
-          std::cout << "\nInserting packet number " << seq_num
-              << " timestamp " << ts << std::endl;
-        } else {
-          std::cout << "\nLost packet number " << seq_num
-              << " timestamp " << ts << std::endl;
-        }
-      }
-    }
-  }
-
-  void TearDown() {
-    delete channel_;
-
-    fclose(seq_num_fid_);
-    fclose(send_ts_fid_);
-    fclose(receive_ts_fid_);
-    fclose(pcm_out_fid_);
-    pcm_in_fid_.Close();
-  }
-
-  ~InsertPacketWithTiming() {
-    delete sender_clock_;
-    delete receiver_clock_;
-  }
-
-  // Are there more info to simulate.
-  bool HasPackets() {
-    if (feof(seq_num_fid_) || feof(send_ts_fid_) || feof(receive_ts_fid_))
-      return false;
-    return true;
-  }
-
-  // Jitter buffer delay.
-  void Delay(int* optimal_delay, int* current_delay) {
-    NetworkStatistics statistics;
-    receive_acm_->GetNetworkStatistics(&statistics);
-    *optimal_delay = statistics.preferredBufferSize;
-    *current_delay = statistics.currentBufferSize;
-  }
-
- private:
-  uint32_t SendTimestamp() {
-    uint32_t t;
-    EXPECT_EQ(1, fscanf(send_ts_fid_, "%u\n", &t));
-    return t;
-  }
-
-  uint32_t ReceiveTimestamp() {
-    uint32_t t;
-    EXPECT_EQ(1, fscanf(receive_ts_fid_, "%u\n", &t));
-    return t;
-  }
-
-  int SequenceNumber() {
-    int n;
-    EXPECT_EQ(1, fscanf(seq_num_fid_, "%d\n", &n));
-    return n;
-  }
-
-  // This class just creates these pointers, not deleting them. They are deleted
-  // by the associated ACM.
-  SimulatedClock* sender_clock_;
-  SimulatedClock* receiver_clock_;
-
-  rtc::scoped_ptr<AudioCodingModule> send_acm_;
-  rtc::scoped_ptr<AudioCodingModule> receive_acm_;
-  Channel* channel_;
-
-  FILE* seq_num_fid_;  // Input (text), one sequence number per line.
-  FILE* send_ts_fid_;  // Input (text), one send timestamp per line.
-  FILE* receive_ts_fid_;  // Input (text), one receive timestamp per line.
-  FILE* pcm_out_fid_;  // Output PCM16.
-
-  PCMFile pcm_in_fid_;  // Input PCM16.
-
-  int samples_in_1ms_;
-
-  // TODO(turajs): this can be computed from the send timestamp, but there is
-  // some complication to account for lost and reordered packets.
-  int num_10ms_in_codec_frame_;
-
-  float time_to_insert_packet_ms_;
-  uint32_t next_receive_ts_;
-  uint32_t time_to_playout_audio_ms_;
-
-  AudioFrame frame_;
-
-  double loss_threshold_;
-
-  // Output (text), sequence number, playout timestamp, time (ms) of playout,
-  // per line.
-  FILE* playout_timing_fid_;
-};
-
-}  // webrtc
-
-int main(int argc, char* argv[]) {
-  google::ParseCommandLineFlags(&argc, &argv, true);
-  webrtc::InsertPacketWithTiming test;
-  test.SetUp();
-
-  FILE* delay_log = NULL;
-  if (FLAGS_delay.size() > 0) {
-    delay_log = fopen(FLAGS_delay.c_str(), "wt");
-    if (delay_log == NULL) {
-      std::cout << "Cannot open the file to log delay values." << std::endl;
-      exit(1);
-    }
-  }
-
-  uint32_t action_taken;
-  int optimal_delay_ms;
-  int current_delay_ms;
-  while (test.HasPackets()) {
-    test.TickOneMillisecond(&action_taken);
-
-    if (action_taken != 0) {
-      test.Delay(&optimal_delay_ms, &current_delay_ms);
-      if (delay_log != NULL) {
-        fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
-      }
-    }
-  }
-  std::cout << std::endl;
-  test.TearDown();
-  if (delay_log != NULL)
-    fclose(delay_log);
-}