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..ba92b5662653a21f16b5c626a7637696a5c32898 |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc |
@@ -0,0 +1,86 @@ |
+/* |
+ * 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/array_view.h" |
+#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(std::initializer_list<Timing::Turn> il) : Timing() { |
+ for (auto turn : il) { |
+ AppendTurn(turn); |
+ } |
kwiberg-webrtc
2017/03/21 22:35:15
Can't you pass the initializer_list directly to tu
AleBzk
2017/03/22 10:07:22
Acknowledged.
|
+} |
+ |
+Timing::~Timing() {} |
+ |
+void Timing::Clear() { |
+ // TODO(alessiob): check if pointers to Turn instances must be explicitly |
+ // deleted (I assume not). |
kwiberg-webrtc
2017/03/21 22:35:15
Comment no longer applicable.
AleBzk
2017/03/22 10:07:22
Acknowledged.
|
+ turns_.clear(); |
+} |
+ |
+void Timing::AppendTurn(Timing::Turn turn) { |
+ turns_.push_back(turn); |
+} |
+ |
+void Timing::Load(const std::string& timing_filepath) { |
+ // Line parser. |
+ auto parse_line_ = [](const std::string& line) { |
+ std::vector<std::string> fields; |
+ rtc::split(line, ' ', &fields); |
+ RTC_CHECK_EQ(fields.size(), 3); |
+ return Timing::Turn(fields[0], fields[1], std::atol(fields[2].c_str())); |
+ }; |
+ |
+ // Parse lines. |
+ std::string line; |
+ std::ifstream infile(timing_filepath); |
+ while (std::getline(infile, line)) { |
+ if (line.empty()) |
+ continue; |
+ AppendTurn(parse_line_(line)); |
+ } |
+ 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(); |
+} |
+ |
+rtc::ArrayView<const Timing::Turn> Timing::turns() const { |
+ return turns_; |
+} |
+ |
+} // namespace conversational_speech |
+} // namespace test |
+} // namespace webrtc |