Source code
Revision control
Copy as Markdown
Other Tools
/*
* 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/multiend_call.h"
#include <algorithm>
#include <iterator>
#include "absl/strings/string_view.h"
#include "rtc_base/logging.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace test {
namespace conversational_speech {
MultiEndCall::MultiEndCall(
rtc::ArrayView<const Turn> timing,
absl::string_view audiotracks_path,
std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory)
: timing_(timing),
audiotracks_path_(audiotracks_path),
wavreader_abstract_factory_(std::move(wavreader_abstract_factory)),
valid_(false) {
FindSpeakerNames();
if (CreateAudioTrackReaders())
valid_ = CheckTiming();
}
MultiEndCall::~MultiEndCall() = default;
void MultiEndCall::FindSpeakerNames() {
RTC_DCHECK(speaker_names_.empty());
for (const Turn& turn : timing_) {
speaker_names_.emplace(turn.speaker_name);
}
}
bool MultiEndCall::CreateAudioTrackReaders() {
RTC_DCHECK(audiotrack_readers_.empty());
sample_rate_hz_ = 0; // Sample rate will be set when reading the first track.
for (const Turn& turn : timing_) {
auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
if (it != audiotrack_readers_.end())
continue;
const std::string audiotrack_file_path =
test::JoinFilename(audiotracks_path_, turn.audiotrack_file_name);
// Map the audiotrack file name to a new instance of WavReaderInterface.
std::unique_ptr<WavReaderInterface> wavreader =
wavreader_abstract_factory_->Create(
test::JoinFilename(audiotracks_path_, turn.audiotrack_file_name));
if (sample_rate_hz_ == 0) {
sample_rate_hz_ = wavreader->SampleRate();
} else if (sample_rate_hz_ != wavreader->SampleRate()) {
RTC_LOG(LS_ERROR)
<< "All the audio tracks should have the same sample rate.";
return false;
}
if (wavreader->NumChannels() != 1) {
RTC_LOG(LS_ERROR) << "Only mono audio tracks supported.";
return false;
}
audiotrack_readers_.emplace(turn.audiotrack_file_name,
std::move(wavreader));
}
return true;
}
bool MultiEndCall::CheckTiming() {
struct Interval {
size_t begin;
size_t end;
};
size_t number_of_turns = timing_.size();
auto millisecond_to_samples = [](int ms, int sr) -> int {
// Truncation may happen if the sampling rate is not an integer multiple
// of 1000 (e.g., 44100).
return ms * sr / 1000;
};
auto in_interval = [](size_t value, const Interval& interval) {
return interval.begin <= value && value < interval.end;
};
total_duration_samples_ = 0;
speaking_turns_.clear();
// Begin and end timestamps for the last two turns (unit: number of samples).
Interval second_last_turn = {0, 0};
Interval last_turn = {0, 0};
// Initialize map to store speaking turn indices of each speaker (used to
// detect self cross-talk).
std::map<std::string, std::vector<size_t>> speaking_turn_indices;
for (const std::string& speaker_name : speaker_names_) {
speaking_turn_indices.emplace(std::piecewise_construct,
std::forward_as_tuple(speaker_name),
std::forward_as_tuple());
}
// Parse turns.
for (size_t turn_index = 0; turn_index < number_of_turns; ++turn_index) {
const Turn& turn = timing_[turn_index];
auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
RTC_CHECK(it != audiotrack_readers_.end())
<< "Audio track reader not created";
// Begin and end timestamps for the current turn.
int offset_samples =
millisecond_to_samples(turn.offset, it->second->SampleRate());
std::size_t begin_timestamp = last_turn.end + offset_samples;
std::size_t end_timestamp = begin_timestamp + it->second->NumSamples();
RTC_LOG(LS_INFO) << "turn #" << turn_index << " " << begin_timestamp << "-"
<< end_timestamp << " ms";
// The order is invalid if the offset is negative and its absolute value is
// larger then the duration of the previous turn.
if (offset_samples < 0 &&
-offset_samples > static_cast<int>(last_turn.end - last_turn.begin)) {
RTC_LOG(LS_ERROR) << "invalid order";
return false;
}
// Cross-talk with 3 or more speakers occurs when the beginning of the
// current interval falls in the last two turns.
if (turn_index > 1 && in_interval(begin_timestamp, last_turn) &&
in_interval(begin_timestamp, second_last_turn)) {
RTC_LOG(LS_ERROR) << "cross-talk with 3+ speakers";
return false;
}
// Append turn.
speaking_turns_.emplace_back(turn.speaker_name, turn.audiotrack_file_name,
begin_timestamp, end_timestamp, turn.gain);
// Save speaking turn index for self cross-talk detection.
RTC_DCHECK_EQ(speaking_turns_.size(), turn_index + 1);
speaking_turn_indices[turn.speaker_name].push_back(turn_index);
// Update total duration of the consversational speech.
if (total_duration_samples_ < end_timestamp)
total_duration_samples_ = end_timestamp;
// Update and continue with next turn.
second_last_turn = last_turn;
last_turn.begin = begin_timestamp;
last_turn.end = end_timestamp;
}
// Detect self cross-talk.
for (const std::string& speaker_name : speaker_names_) {
RTC_LOG(LS_INFO) << "checking self cross-talk for <" << speaker_name << ">";
// Copy all turns for this speaker to new vector.
std::vector<SpeakingTurn> speaking_turns_for_name;
std::copy_if(speaking_turns_.begin(), speaking_turns_.end(),
std::back_inserter(speaking_turns_for_name),
[&speaker_name](const SpeakingTurn& st) {
return st.speaker_name == speaker_name;
});
// Check for overlap between adjacent elements.
// This is a sufficient condition for self cross-talk since the intervals
// are sorted by begin timestamp.
auto overlap = std::adjacent_find(
speaking_turns_for_name.begin(), speaking_turns_for_name.end(),
[](const SpeakingTurn& a, const SpeakingTurn& b) {
return a.end > b.begin;
});
if (overlap != speaking_turns_for_name.end()) {
RTC_LOG(LS_ERROR) << "Self cross-talk detected";
return false;
}
}
return true;
}
} // namespace conversational_speech
} // namespace test
} // namespace webrtc