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/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
#include <stddef.h>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
#include "webrtc/rtc_base/array_view.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
namespace test {
namespace conversational_speech {
class MultiEndCall {
public:
struct SpeakingTurn {
// Constructor required in order to use std::vector::emplace_back().
SpeakingTurn(std::string new_speaker_name,
std::string new_audiotrack_file_name,
size_t new_begin, size_t new_end)
: speaker_name(std::move(new_speaker_name)),
audiotrack_file_name(std::move(new_audiotrack_file_name)),
begin(new_begin), end(new_end) {}
std::string speaker_name;
std::string audiotrack_file_name;
size_t begin;
size_t end;
};
MultiEndCall(
rtc::ArrayView<const Turn> timing, const std::string& audiotracks_path,
std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
~MultiEndCall();
const std::set<std::string>& speaker_names() const { return speaker_names_; }
const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
audiotrack_readers() const { return audiotrack_readers_; }
bool valid() const { return valid_; }
int sample_rate() const { return sample_rate_hz_; }
size_t total_duration_samples() const { return total_duration_samples_; }
const std::vector<SpeakingTurn>& speaking_turns() const {
return speaking_turns_; }
private:
// Finds unique speaker names.
void FindSpeakerNames();
// Creates one WavReader instance for each unique audiotrack. It returns false
// if the audio tracks do not have the same sample rate or if they are not
// mono.
bool CreateAudioTrackReaders();
// Validates the speaking turns timing information. Accepts cross-talk, but
// only up to 2 speakers. Rejects unordered turns and self cross-talk.
bool CheckTiming();
rtc::ArrayView<const Turn> timing_;
const std::string& audiotracks_path_;
std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory_;
std::set<std::string> speaker_names_;
std::map<std::string, std::unique_ptr<WavReaderInterface>>
audiotrack_readers_;
bool valid_;
int sample_rate_hz_;
size_t total_duration_samples_;
std::vector<SpeakingTurn> speaking_turns_;
RTC_DISALLOW_COPY_AND_ASSIGN(MultiEndCall);
};
} // namespace conversational_speech
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_