Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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modules/audio_coding/test/insert_packet_with_timing.cc
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modules/audio_coding/test/insert_packet_with_timing.cc
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/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdio.h>
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#include <string.h>
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#include <memory>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
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#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
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#include "webrtc/modules/audio_coding/test/Channel.h"
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#include "webrtc/modules/audio_coding/test/PCMFile.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/rtc_base/flags.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/test/gtest.h"
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#include "webrtc/test/testsupport/fileutils.h"
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// Codec.
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DEFINE_string(codec, "opus", "Codec Name");
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DEFINE_int(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
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DEFINE_int(codec_channels, 1, "Number of channels of the codec.");
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// PCM input/output.
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DEFINE_string(input, "", "Input PCM file at 16 kHz.");
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DEFINE_bool(input_stereo, false, "Input is stereo.");
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DEFINE_int(input_fs_hz, 32000, "Input sample rate Hz.");
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DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
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DEFINE_int(output_fs_hz, 32000, "Output sample rate Hz");
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// Timing files
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DEFINE_string(seq_num, "seq_num", "Sequence number file.");
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DEFINE_string(send_ts, "send_timestamp", "Send timestamp file.");
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DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
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// Delay logging
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DEFINE_string(delay, "", "Log for delay.");
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// Other setups
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DEFINE_bool(verbose, false, "Verbosity.");
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DEFINE_float(loss_rate, 0, "Rate of packet loss < 1");
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DEFINE_bool(help, false, "Prints this message.");
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const int32_t kAudioPlayedOut = 0x00000001;
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const int32_t kPacketPushedIn = 0x00000001 << 1;
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const int kPlayoutPeriodMs = 10;
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namespace webrtc {
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class InsertPacketWithTiming {
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public:
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InsertPacketWithTiming()
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: sender_clock_(new SimulatedClock(0)),
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receiver_clock_(new SimulatedClock(0)),
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send_acm_(AudioCodingModule::Create(0, sender_clock_)),
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receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
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channel_(new Channel),
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seq_num_fid_(fopen(FLAG_seq_num, "rt")),
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send_ts_fid_(fopen(FLAG_send_ts, "rt")),
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receive_ts_fid_(fopen(FLAG_receive_ts, "rt")),
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pcm_out_fid_(fopen(FLAG_output, "wb")),
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samples_in_1ms_(48),
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num_10ms_in_codec_frame_(2), // Typical 20 ms frames.
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time_to_insert_packet_ms_(3), // An arbitrary offset on pushing packet.
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next_receive_ts_(0),
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time_to_playout_audio_ms_(kPlayoutPeriodMs),
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loss_threshold_(0),
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playout_timing_fid_(fopen("playout_timing.txt", "wt")) {}
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void SetUp() {
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ASSERT_TRUE(sender_clock_ != NULL);
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ASSERT_TRUE(receiver_clock_ != NULL);
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ASSERT_TRUE(send_acm_.get() != NULL);
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ASSERT_TRUE(receive_acm_.get() != NULL);
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ASSERT_TRUE(channel_ != NULL);
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ASSERT_TRUE(seq_num_fid_ != NULL);
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ASSERT_TRUE(send_ts_fid_ != NULL);
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ASSERT_TRUE(receive_ts_fid_ != NULL);
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ASSERT_TRUE(playout_timing_fid_ != NULL);
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next_receive_ts_ = ReceiveTimestamp();
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CodecInst codec;
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ASSERT_EQ(0, AudioCodingModule::Codec(FLAG_codec, &codec,
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FLAG_codec_sample_rate_hz,
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FLAG_codec_channels));
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ASSERT_EQ(0, receive_acm_->InitializeReceiver());
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ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
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ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
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CodecInstToSdp(codec)));
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// Set codec-dependent parameters.
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samples_in_1ms_ = codec.plfreq / 1000;
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num_10ms_in_codec_frame_ = codec.pacsize / (codec.plfreq / 100);
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channel_->RegisterReceiverACM(receive_acm_.get());
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send_acm_->RegisterTransportCallback(channel_);
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if (strlen(FLAG_input) == 0) {
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std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
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"pcm");
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pcm_in_fid_.Open(file_name, 32000, "r", true); // auto-rewind
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std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
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} else {
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pcm_in_fid_.Open(FLAG_input, static_cast<uint16_t>(FLAG_input_fs_hz),
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"r", true); // auto-rewind
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std::cout << "Input file " << FLAG_input << "at " << FLAG_input_fs_hz
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<< " Hz in " << ((FLAG_input_stereo) ? "stereo." : "mono.")
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<< std::endl;
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pcm_in_fid_.ReadStereo(FLAG_input_stereo);
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}
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ASSERT_TRUE(pcm_out_fid_ != NULL);
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std::cout << "Output file " << FLAG_output << " at " << FLAG_output_fs_hz
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<< " Hz." << std::endl;
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// Other setups
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if (FLAG_loss_rate > 0)
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loss_threshold_ = RAND_MAX * FLAG_loss_rate;
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else
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loss_threshold_ = 0;
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}
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void TickOneMillisecond(uint32_t* action) {
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// One millisecond passed.
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time_to_insert_packet_ms_--;
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time_to_playout_audio_ms_--;
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sender_clock_->AdvanceTimeMilliseconds(1);
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receiver_clock_->AdvanceTimeMilliseconds(1);
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// Reset action.
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*action = 0;
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// Is it time to pull audio?
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if (time_to_playout_audio_ms_ == 0) {
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time_to_playout_audio_ms_ = kPlayoutPeriodMs;
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bool muted;
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receive_acm_->PlayoutData10Ms(static_cast<int>(FLAG_output_fs_hz),
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&frame_, &muted);
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ASSERT_FALSE(muted);
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fwrite(frame_.data(), sizeof(*frame_.data()),
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frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
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*action |= kAudioPlayedOut;
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}
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// Is it time to push in next packet?
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if (time_to_insert_packet_ms_ <= .5) {
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*action |= kPacketPushedIn;
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// Update time-to-insert packet.
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uint32_t t = next_receive_ts_;
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next_receive_ts_ = ReceiveTimestamp();
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time_to_insert_packet_ms_ += static_cast<float>(next_receive_ts_ - t) /
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samples_in_1ms_;
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// Push in just enough audio.
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for (int n = 0; n < num_10ms_in_codec_frame_; n++) {
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pcm_in_fid_.Read10MsData(frame_);
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EXPECT_GE(send_acm_->Add10MsData(frame_), 0);
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}
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// Set the parameters for the packet to be pushed in receiver ACM right
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// now.
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uint32_t ts = SendTimestamp();
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int seq_num = SequenceNumber();
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bool lost = false;
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channel_->set_send_timestamp(ts);
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channel_->set_sequence_number(seq_num);
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if (loss_threshold_ > 0 && rand() < loss_threshold_) {
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channel_->set_num_packets_to_drop(1);
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lost = true;
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}
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if (FLAG_verbose) {
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if (!lost) {
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std::cout << "\nInserting packet number " << seq_num
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<< " timestamp " << ts << std::endl;
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} else {
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std::cout << "\nLost packet number " << seq_num
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<< " timestamp " << ts << std::endl;
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}
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}
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}
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}
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void TearDown() {
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delete channel_;
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fclose(seq_num_fid_);
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fclose(send_ts_fid_);
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fclose(receive_ts_fid_);
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fclose(pcm_out_fid_);
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pcm_in_fid_.Close();
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}
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~InsertPacketWithTiming() {
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delete sender_clock_;
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delete receiver_clock_;
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}
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// Are there more info to simulate.
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bool HasPackets() {
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if (feof(seq_num_fid_) || feof(send_ts_fid_) || feof(receive_ts_fid_))
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return false;
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return true;
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}
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// Jitter buffer delay.
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void Delay(int* optimal_delay, int* current_delay) {
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NetworkStatistics statistics;
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receive_acm_->GetNetworkStatistics(&statistics);
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*optimal_delay = statistics.preferredBufferSize;
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*current_delay = statistics.currentBufferSize;
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}
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private:
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uint32_t SendTimestamp() {
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uint32_t t;
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EXPECT_EQ(1, fscanf(send_ts_fid_, "%u\n", &t));
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return t;
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}
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uint32_t ReceiveTimestamp() {
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uint32_t t;
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EXPECT_EQ(1, fscanf(receive_ts_fid_, "%u\n", &t));
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return t;
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}
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int SequenceNumber() {
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int n;
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EXPECT_EQ(1, fscanf(seq_num_fid_, "%d\n", &n));
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return n;
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}
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// This class just creates these pointers, not deleting them. They are deleted
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// by the associated ACM.
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SimulatedClock* sender_clock_;
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SimulatedClock* receiver_clock_;
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std::unique_ptr<AudioCodingModule> send_acm_;
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std::unique_ptr<AudioCodingModule> receive_acm_;
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Channel* channel_;
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FILE* seq_num_fid_; // Input (text), one sequence number per line.
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FILE* send_ts_fid_; // Input (text), one send timestamp per line.
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FILE* receive_ts_fid_; // Input (text), one receive timestamp per line.
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FILE* pcm_out_fid_; // Output PCM16.
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PCMFile pcm_in_fid_; // Input PCM16.
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int samples_in_1ms_;
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// TODO(turajs): this can be computed from the send timestamp, but there is
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// some complication to account for lost and reordered packets.
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int num_10ms_in_codec_frame_;
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float time_to_insert_packet_ms_;
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uint32_t next_receive_ts_;
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uint32_t time_to_playout_audio_ms_;
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AudioFrame frame_;
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double loss_threshold_;
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// Output (text), sequence number, playout timestamp, time (ms) of playout,
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// per line.
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FILE* playout_timing_fid_;
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};
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} // webrtc
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int main(int argc, char* argv[]) {
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if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
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return 1;
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}
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if (FLAG_help) {
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rtc::FlagList::Print(nullptr, false);
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return 0;
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}
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webrtc::InsertPacketWithTiming test;
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test.SetUp();
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FILE* delay_log = NULL;
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if (strlen(FLAG_delay) > 0) {
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delay_log = fopen(FLAG_delay, "wt");
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if (delay_log == NULL) {
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std::cout << "Cannot open the file to log delay values." << std::endl;
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exit(1);
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}
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}
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uint32_t action_taken;
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int optimal_delay_ms;
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int current_delay_ms;
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while (test.HasPackets()) {
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test.TickOneMillisecond(&action_taken);
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if (action_taken != 0) {
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test.Delay(&optimal_delay_ms, ¤t_delay_ms);
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if (delay_log != NULL) {
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fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
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}
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}
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}
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std::cout << std::endl;
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test.TearDown();
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if (delay_log != NULL)
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fclose(delay_log);
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}
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