Chromium Code Reviews| Index: webrtc/modules/audio_processing/test/conversational_speech/timing.cc | 
| diff --git a/webrtc/modules/audio_processing/test/conversational_speech/timing.cc b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc | 
| new file mode 100644 | 
| index 0000000000000000000000000000000000000000..83c99047ac1c726c5931808fa1d9bee5f1350f57 | 
| --- /dev/null | 
| +++ b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc | 
| @@ -0,0 +1,89 @@ | 
| +/* | 
| + * Copyright (c) 2017 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 "webrtc/modules/audio_processing/test/conversational_speech/timing.h" | 
| + | 
| +#include <fstream> | 
| +#include <iostream> | 
| +#include <utility> | 
| + | 
| +#include "webrtc/base/stringencode.h" | 
| + | 
| +namespace webrtc { | 
| +namespace test { | 
| +namespace conversational_speech { | 
| + | 
| +bool Timing::Turn::operator ==(const Turn &b) const { | 
| + return b.speaker_name == speaker_name && | 
| + b.audiotrack_file_name == audiotrack_file_name && | 
| + b.offset == offset; | 
| +} | 
| + | 
| +Timing::Timing() {} | 
| + | 
| +Timing::~Timing() { | 
| + Clear(); | 
| 
 
kwiberg-webrtc
2017/03/16 14:10:08
Not necessary, right?
 
AleBzk
2017/03/20 14:25:28
Acknowledged.
 
 | 
| +} | 
| + | 
| +void Timing::Clear() { | 
| + // TODO(alessiob): check if pointers to Turn instances must be explicitly | 
| + // deleted (I assume not). | 
| 
 
kwiberg-webrtc
2017/03/16 14:10:09
No, unique_ptr will delete the thing it owns---tha
 
AleBzk
2017/03/20 14:25:28
Acknowledged.
 
 | 
| + turns_.clear(); | 
| +} | 
| + | 
| +void Timing::AppendTurn(std::unique_ptr<Timing::Turn> turn) { | 
| + turns_.push_back(std::move(turn)); | 
| +} | 
| + | 
| +void Timing::Load(const std::string& timing_filepath) { | 
| + std::string line; | 
| + | 
| + // Line parser. | 
| + auto parse_line_ = [&line]() { | 
| + std::vector<std::string> fields; | 
| + rtc::split(line, ' ', &fields); | 
| + RTC_CHECK_EQ(fields.size(), 3); | 
| + return std::unique_ptr<Timing::Turn>(new Turn( | 
| + fields[0], fields[1], std::atol(fields[2].c_str()))); | 
| + }; | 
| + | 
| + // Parse lines. | 
| + std::ifstream infile(timing_filepath); | 
| + while (std::getline(infile, line)) { | 
| + if (line.empty()) | 
| + continue; | 
| + AppendTurn(parse_line_()); | 
| 
 
kwiberg-webrtc
2017/03/16 14:10:09
Pass line as an argument here, instead of capturin
 
AleBzk
2017/03/20 14:25:28
Done.
 
 | 
| + } | 
| + infile.close(); | 
| +} | 
| + | 
| +void Timing::Save(const std::string& timing_filepath) const { | 
| + std::ofstream outfile(timing_filepath); | 
| + // TODO(alessio): check if file open for writing. | 
| + for (const auto& turn : turns_) { | 
| + outfile << turn->speaker_name << " " << turn->audiotrack_file_name | 
| + << " " << turn->offset << std::endl; | 
| + } | 
| + outfile.close(); | 
| +} | 
| + | 
| +const std::vector<std::unique_ptr<Timing::Turn>>& Timing::turns() const { | 
| + // TODO(alessiob): not sure if this is the best way to return, maybe we can | 
| + // return an iterator. | 
| 
 
kwiberg-webrtc
2017/03/16 14:10:09
This is fine. You could also return an ArrayView<s
 
AleBzk
2017/03/20 14:25:28
I'm new with ArrayView, I guess you refer to the o
 
kwiberg-webrtc
2017/03/21 22:35:14
Yes. (I wrote it, so if you have questions or comp
 
 | 
| + return turns_; | 
| +} | 
| + | 
| +std::size_t Timing::Size() const { | 
| + return turns_.size(); | 
| +} | 
| 
 
kwiberg-webrtc
2017/03/16 14:10:08
Is this necessary? Callers can do .turns().size().
 
AleBzk
2017/03/20 14:25:28
Removed
 
 | 
| + | 
| +} // namespace conversational_speech | 
| +} // namespace test | 
| +} // namespace webrtc |