2015-09-10 19:31:25 +00:00
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#include <pocketsphinx.h>
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#include <stdexcept>
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#include <fstream>
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#include <memory>
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2015-09-28 19:19:39 +00:00
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#include <vector>
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#include <iostream>
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#include <chrono>
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#include "audio_input/16kHzMonoStream.h"
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2015-09-10 18:05:05 +00:00
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2015-09-10 19:31:25 +00:00
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using std::runtime_error;
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using std::shared_ptr;
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2015-09-28 19:19:39 +00:00
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using std::unique_ptr;
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2015-09-10 18:05:05 +00:00
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2015-09-10 19:31:25 +00:00
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#define MODELDIR "X:/dev/projects/LipSync/lib/pocketsphinx/model"
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2015-09-28 19:19:39 +00:00
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// Converts a float in the range -1..1 to a signed 16-bit int
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int16_t floatSampleToInt16(float sample) {
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sample = std::max(sample, -1.0f);
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sample = std::min(sample, 1.0f);
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return static_cast<int16_t>(((sample + 1) / 2) * (INT16_MAX - INT16_MIN) + INT16_MIN);
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}
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2015-09-10 19:31:25 +00:00
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int main(int argc, char *argv[]) {
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shared_ptr<cmd_ln_t> config(
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cmd_ln_init(
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nullptr, ps_args(), true,
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// Set acoustic model
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"-hmm", MODELDIR "/en-us/en-us",
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// Set phonetic language model
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"-allphone", MODELDIR "/en-us/en-us-phone.lm.bin",
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// The following settings are Voodoo to me.
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// I copied them from http://cmusphinx.sourceforge.net/wiki/phonemerecognition
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// Set beam width applied to every frame in Viterbi search
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"-beam", "1e-20",
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// Set beam width applied to phone transitions
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"-pbeam", "1e-20",
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// Set language model probability weight
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"-lw", "2.0",
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nullptr),
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[](cmd_ln_t* config) { cmd_ln_free_r(config); });
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if (!config) throw runtime_error("Error creating configuration.");
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shared_ptr<ps_decoder_t> recognizer(
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ps_init(config.get()),
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[](ps_decoder_t* recognizer) { ps_free(recognizer); });
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if (!recognizer) throw runtime_error("Error creating speech recognizer.");
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2015-09-28 19:19:39 +00:00
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unique_ptr<AudioStream> audioStream =
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create16kHzMonoStream(R"(C:\Users\Daniel\Desktop\audio-test\test 16000Hz 1ch 16bit.wav)");
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2015-09-10 19:31:25 +00:00
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int error = ps_start_utt(recognizer.get());
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if (error) throw runtime_error("Error starting utterance processing.");
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2015-09-28 19:19:39 +00:00
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auto start = std::chrono::steady_clock::now();
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std::vector<int16_t> buffer;
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const int capacity = 1600;
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buffer.reserve(capacity); // 0.1 second capacity
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int sampleCount = 0;
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do {
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// Read to buffer
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buffer.clear();
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while (buffer.size() < capacity) {
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float sample;
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if (!audioStream->getNextSample(sample)) break;
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buffer.push_back(floatSampleToInt16(sample));
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}
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// Analyze buffer
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int searchedFrameCount = ps_process_raw(recognizer.get(), buffer.data(), buffer.size(), false, false);
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2015-09-10 19:31:25 +00:00
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if (searchedFrameCount < 0) throw runtime_error("Error decoding raw audio data.");
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2015-09-28 19:19:39 +00:00
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sampleCount += buffer.size();
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std::cout << sampleCount / 16000.0 << "s\n";
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} while (buffer.size());
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2015-09-10 19:31:25 +00:00
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error = ps_end_utt(recognizer.get());
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if (error) throw runtime_error("Error ending utterance processing.");
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2015-09-28 19:19:39 +00:00
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auto end = std::chrono::steady_clock::now();
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std::cout << std::chrono::duration_cast<std::chrono::duration<double>>(end - start).count() << "\n";
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2015-09-10 19:31:25 +00:00
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ps_seg_t *segmentationIter;
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int32 score;
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for (segmentationIter = ps_seg_iter(recognizer.get(), &score); segmentationIter; segmentationIter = ps_seg_next(segmentationIter)) {
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// Get phoneme
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char const *phoneme = ps_seg_word(segmentationIter);
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// Get timing
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int startFrame, endFrame;
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ps_seg_frames(segmentationIter, &startFrame, &endFrame);
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printf(">>> %-5s %-5d %-5d\n", phoneme, startFrame, endFrame);
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}
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return 0;
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2015-09-10 18:05:05 +00:00
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}
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