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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc | 73 |
1 files changed, 73 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc new file mode 100644 index 0000000000..95ec9f542e --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc @@ -0,0 +1,73 @@ +/* + * 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 "modules/audio_processing/test/conversational_speech/timing.h" + +#include <fstream> +#include <iostream> +#include <string> + +#include "absl/strings/string_view.h" +#include "rtc_base/string_encode.h" + +namespace webrtc { +namespace test { +namespace conversational_speech { + +bool Turn::operator==(const Turn& b) const { + return b.speaker_name == speaker_name && + b.audiotrack_file_name == audiotrack_file_name && b.offset == offset && + b.gain == gain; +} + +std::vector<Turn> LoadTiming(absl::string_view timing_filepath) { + // Line parser. + auto parse_line = [](absl::string_view line) { + std::vector<absl::string_view> fields = rtc::split(line, ' '); + RTC_CHECK_GE(fields.size(), 3); + RTC_CHECK_LE(fields.size(), 4); + int gain = 0; + if (fields.size() == 4) { + gain = rtc::StringToNumber<int>(fields[3]).value_or(0); + } + return Turn(fields[0], fields[1], + rtc::StringToNumber<int>(fields[2]).value_or(0), gain); + }; + + // Init. + std::vector<Turn> timing; + + // Parse lines. + std::string line; + std::ifstream infile(std::string{timing_filepath}); + while (std::getline(infile, line)) { + if (line.empty()) + continue; + timing.push_back(parse_line(line)); + } + infile.close(); + + return timing; +} + +void SaveTiming(absl::string_view timing_filepath, + rtc::ArrayView<const Turn> timing) { + std::ofstream outfile(std::string{timing_filepath}); + RTC_CHECK(outfile.is_open()); + for (const Turn& turn : timing) { + outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " " + << turn.offset << " " << turn.gain << std::endl; + } + outfile.close(); +} + +} // namespace conversational_speech +} // namespace test +} // namespace webrtc |