mirror of https://github.com/rhasspy/piper
First working version with libpiper_phonemize
Michael Hansen
12 months ago
parent
7d27863b48
commit
810fad44cf
Binary file not shown.
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#ifndef CONFIG_H_
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#define CONFIG_H_
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#include <filesystem>
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#include <map>
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#include <optional>
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#include <set>
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#include <stdexcept>
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#include <string>
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#include <vector>
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#include "json.hpp"
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#include "utf8.h"
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using namespace std;
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using json = nlohmann::json;
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namespace piper {
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typedef char32_t Phoneme;
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typedef int64_t PhonemeId;
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typedef int64_t SpeakerId;
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const string DefaultVoice = "en-us";
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enum eSpeakMode { Text, TextWithPhonemes, SSML };
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struct eSpeakConfig {
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string voice = DefaultVoice;
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eSpeakMode mode = Text;
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// Characters that eSpeak uses to break apart paragraphs/sentences
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set<Phoneme> clauseBreakers{U'.', U'?', U'!', U',', U';', U':'};
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Phoneme fullStop = U'.';
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Phoneme comma = U',';
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Phoneme question = U'?';
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Phoneme exclamation = U'!';
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};
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struct PhonemizeConfig {
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optional<map<Phoneme, vector<Phoneme>>> phonemeMap;
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map<Phoneme, vector<PhonemeId>> phonemeIdMap;
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PhonemeId idPad = 0; // padding (optionally interspersed)
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PhonemeId idBos = 1; // beginning of sentence
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PhonemeId idEos = 2; // end of sentence
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bool interspersePad = true;
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optional<eSpeakConfig> eSpeak;
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};
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struct SynthesisConfig {
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float noiseScale = 0.667f;
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float lengthScale = 1.0f;
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float noiseW = 0.8f;
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int sampleRate = 22050;
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int sampleWidth = 2; // 16-bit
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int channels = 1; // mono
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optional<SpeakerId> speakerId;
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float sentenceSilenceSeconds = 0.2f;
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};
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struct ModelConfig {
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int numSpeakers;
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};
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bool isSingleCodepoint(string s) {
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return utf8::distance(s.begin(), s.end()) == 1;
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}
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Phoneme getCodepoint(string s) {
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utf8::iterator character_iter(s.begin(), s.begin(), s.end());
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return *character_iter;
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}
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void parsePhonemizeConfig(json &configRoot, PhonemizeConfig &phonemizeConfig) {
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if (configRoot.contains("espeak")) {
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if (!phonemizeConfig.eSpeak) {
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phonemizeConfig.eSpeak.emplace();
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}
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auto espeakValue = configRoot["espeak"];
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if (espeakValue.contains("voice")) {
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phonemizeConfig.eSpeak->voice = espeakValue["voice"].get<string>();
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}
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}
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// phoneme to [phoneme] map
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if (configRoot.contains("phoneme_map")) {
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if (!phonemizeConfig.phonemeMap) {
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phonemizeConfig.phonemeMap.emplace();
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}
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auto phonemeMapValue = configRoot["phoneme_map"];
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for (auto &fromPhonemeItem : phonemeMapValue.items()) {
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string fromPhoneme = fromPhonemeItem.key();
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if (!isSingleCodepoint(fromPhoneme)) {
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throw runtime_error("Phonemes must be one codepoint (phoneme map)");
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}
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auto fromCodepoint = getCodepoint(fromPhoneme);
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for (auto &toPhonemeValue : fromPhonemeItem.value()) {
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string toPhoneme = toPhonemeValue.get<string>();
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if (!isSingleCodepoint(toPhoneme)) {
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throw runtime_error("Phonemes must be one codepoint (phoneme map)");
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}
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auto toCodepoint = getCodepoint(toPhoneme);
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(*phonemizeConfig.phonemeMap)[fromCodepoint].push_back(toCodepoint);
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}
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}
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}
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// phoneme to [id] map
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if (configRoot.contains("phoneme_id_map")) {
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auto phonemeIdMapValue = configRoot["phoneme_id_map"];
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for (auto &fromPhonemeItem : phonemeIdMapValue.items()) {
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string fromPhoneme = fromPhonemeItem.key();
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if (!isSingleCodepoint(fromPhoneme)) {
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throw runtime_error("Phonemes must be one codepoint (phoneme id map)");
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}
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auto fromCodepoint = getCodepoint(fromPhoneme);
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for (auto &toIdValue : fromPhonemeItem.value()) {
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PhonemeId toId = toIdValue.get<PhonemeId>();
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phonemizeConfig.phonemeIdMap[fromCodepoint].push_back(toId);
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}
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}
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}
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} /* parsePhonemizeConfig */
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void parseSynthesisConfig(json &configRoot, SynthesisConfig &synthesisConfig) {
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if (configRoot.contains("audio")) {
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auto audioValue = configRoot["audio"];
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if (audioValue.contains("sample_rate")) {
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// Default sample rate is 22050 Hz
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synthesisConfig.sampleRate = audioValue.value("sample_rate", 22050);
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}
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}
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} /* parseSynthesisConfig */
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void parseModelConfig(json &configRoot, ModelConfig &modelConfig) {
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modelConfig.numSpeakers = configRoot["num_speakers"].get<SpeakerId>();
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} /* parseModelConfig */
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} // namespace piper
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#endif // CONFIG_H_
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#ifndef MODEL_H_
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#define MODEL_H_
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#include <string>
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#include <onnxruntime_cxx_api.h>
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using namespace std;
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namespace piper {
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const string instanceName{"piper"};
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struct ModelSession {
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Ort::Session onnx;
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Ort::AllocatorWithDefaultOptions allocator;
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Ort::SessionOptions options;
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Ort::Env env;
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ModelSession() : onnx(nullptr){};
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};
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void loadModel(string modelPath, ModelSession &session) {
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session.env = Ort::Env(OrtLoggingLevel::ORT_LOGGING_LEVEL_WARNING,
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instanceName.c_str());
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session.env.DisableTelemetryEvents();
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// Slows down performance by ~2x
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// session.options.SetIntraOpNumThreads(1);
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// Roughly doubles load time for no visible inference benefit
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// session.options.SetGraphOptimizationLevel(
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// GraphOptimizationLevel::ORT_ENABLE_EXTENDED);
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session.options.SetGraphOptimizationLevel(
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GraphOptimizationLevel::ORT_DISABLE_ALL);
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// Slows down performance very slightly
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// session.options.SetExecutionMode(ExecutionMode::ORT_PARALLEL);
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session.options.DisableCpuMemArena();
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session.options.DisableMemPattern();
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session.options.DisableProfiling();
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auto startTime = chrono::steady_clock::now();
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session.onnx = Ort::Session(session.env, filesystem::path(modelPath).c_str(), session.options);
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auto endTime = chrono::steady_clock::now();
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auto loadDuration = chrono::duration<double>(endTime - startTime);
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}
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} // namespace piper
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#endif // MODEL_H_
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#ifndef PHONEMIZE_H_
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#define PHONEMIZE_H_
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#include <filesystem>
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#include <iostream>
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#include <map>
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#include <optional>
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#include <set>
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#include <stdexcept>
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#include <string>
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#include <vector>
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#include <espeak-ng/speak_lib.h>
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#include "config.hpp"
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#include "utf8.h"
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#define CLAUSE_INTONATION_FULL_STOP 0x00000000
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#define CLAUSE_INTONATION_COMMA 0x00001000
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#define CLAUSE_INTONATION_QUESTION 0x00002000
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#define CLAUSE_INTONATION_EXCLAMATION 0x00003000
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#define CLAUSE_TYPE_SENTENCE 0x00080000
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using namespace std;
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namespace piper {
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// Text to phonemes using eSpeak-ng
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void phonemize(string text, PhonemizeConfig &phonemizeConfig,
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vector<vector<Phoneme>> &phonemes) {
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if (!phonemizeConfig.eSpeak) {
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throw runtime_error("Missing eSpeak config");
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}
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auto voice = phonemizeConfig.eSpeak->voice;
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int result = espeak_SetVoiceByName(voice.c_str());
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if (result != 0) {
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throw runtime_error("Failed to set eSpeak-ng voice");
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}
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// Modified by eSpeak
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string textCopy(text);
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utf8::iterator textIter(textCopy.begin(), textCopy.begin(), textCopy.end());
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utf8::iterator textIterEnd(textCopy.end(), textCopy.begin(), textCopy.end());
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vector<char32_t> textClauseBreakers;
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// Identify clause breakers in the sentence, since eSpeak removes them during
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// phonemization.
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//
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// This will unfortunately do the wrong thing with abbreviations, etc.
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while (textIter != textIterEnd) {
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auto codepoint = *textIter;
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if (phonemizeConfig.eSpeak->clauseBreakers.contains(codepoint)) {
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textClauseBreakers.push_back(codepoint);
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}
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textIter++;
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}
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vector<Phoneme> *sentencePhonemes = nullptr;
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const char *inputTextPointer = textCopy.c_str();
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int terminator = 0;
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while (inputTextPointer != NULL) {
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// Modified espeak-ng API to get access to clause terminator
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string clausePhonemes(
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espeak_TextToPhonemes2((const void **)&inputTextPointer,
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/*textmode*/ espeakCHARS_AUTO,
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/*phonememode = IPA*/ 0x02,
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&terminator));
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utf8::iterator phonemeIter(clausePhonemes.begin(), clausePhonemes.begin(),
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clausePhonemes.end());
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utf8::iterator phonemeEnd(clausePhonemes.end(), clausePhonemes.begin(),
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clausePhonemes.end());
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if (!sentencePhonemes) {
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// Start new sentence
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phonemes.emplace_back();
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sentencePhonemes = &phonemes[phonemes.size() - 1];
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}
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sentencePhonemes->insert(sentencePhonemes->end(), phonemeIter, phonemeEnd);
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// Add appropriate puntuation depending on terminator type
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int intonation = terminator & 0x0000F000;
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if (intonation == CLAUSE_INTONATION_FULL_STOP) {
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sentencePhonemes->push_back(phonemizeConfig.eSpeak->fullStop);
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} else if (intonation == CLAUSE_INTONATION_COMMA) {
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sentencePhonemes->push_back(phonemizeConfig.eSpeak->comma);
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} else if (intonation == CLAUSE_INTONATION_QUESTION) {
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sentencePhonemes->push_back(phonemizeConfig.eSpeak->question);
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} else if (intonation == CLAUSE_INTONATION_EXCLAMATION) {
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sentencePhonemes->push_back(phonemizeConfig.eSpeak->exclamation);
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}
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if ((terminator & CLAUSE_TYPE_SENTENCE) == CLAUSE_TYPE_SENTENCE) {
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// End of sentence
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sentencePhonemes = nullptr;
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}
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} // while inputTextPointer != NULL
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} /* phonemize */
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// Phonemes to ids using JSON map
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void phonemes2ids(vector<Phoneme> &phonemes, PhonemizeConfig &phonemizeConfig,
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vector<PhonemeId> &phonemeIds) {
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if (phonemes.empty()) {
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throw runtime_error("No phonemes");
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}
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phonemeIds.push_back(phonemizeConfig.idBos);
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if (phonemizeConfig.interspersePad) {
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phonemeIds.push_back(phonemizeConfig.idPad);
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}
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for (auto phoneme = phonemes.begin(); phoneme != phonemes.end(); phoneme++) {
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if (phonemizeConfig.phonemeIdMap.contains(*phoneme)) {
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for (auto id : phonemizeConfig.phonemeIdMap[*phoneme]) {
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phonemeIds.push_back(id);
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if (phonemizeConfig.interspersePad) {
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phonemeIds.push_back(phonemizeConfig.idPad);
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}
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}
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} else {
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string phonemeStr;
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utf8::append(*phoneme, phonemeStr);
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cerr << "[WARN] No id for phoneme: " << phonemeStr << endl;
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}
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}
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phonemeIds.push_back(phonemizeConfig.idEos);
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} /* phonemes2ids */
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} // namespace piper
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#endif // PHONEMIZE_H_
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#include <array>
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#include <chrono>
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#include <fstream>
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#include <limits>
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#include <stdexcept>
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#include <espeak-ng/speak_lib.h>
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#include <onnxruntime_cxx_api.h>
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#include "piper.hpp"
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#include "utf8.h"
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#include "wavfile.hpp"
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namespace piper {
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// Maximum value for 16-bit signed WAV sample
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const float MAX_WAV_VALUE = 32767.0f;
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const std::string instanceName{"piper"};
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bool isSingleCodepoint(std::string s) {
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return utf8::distance(s.begin(), s.end()) == 1;
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}
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Phoneme getCodepoint(std::string s) {
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utf8::iterator character_iter(s.begin(), s.begin(), s.end());
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return *character_iter;
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}
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void parsePhonemizeConfig(json &configRoot, PhonemizeConfig &phonemizeConfig) {
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if (configRoot.contains("espeak")) {
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if (!phonemizeConfig.eSpeak) {
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phonemizeConfig.eSpeak.emplace();
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}
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auto espeakValue = configRoot["espeak"];
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if (espeakValue.contains("voice")) {
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phonemizeConfig.eSpeak->voice = espeakValue["voice"].get<std::string>();
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}
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}
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if (configRoot.contains("phoneme_type")) {
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auto phonemeTypeStr = configRoot["phoneme_type"].get<std::string>();
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if (phonemeTypeStr == "text") {
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phonemizeConfig.phonemeType = TextPhonemes;
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}
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}
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// phoneme to [phoneme] map
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if (configRoot.contains("phoneme_map")) {
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if (!phonemizeConfig.phonemeMap) {
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phonemizeConfig.phonemeMap.emplace();
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}
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auto phonemeMapValue = configRoot["phoneme_map"];
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for (auto &fromPhonemeItem : phonemeMapValue.items()) {
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std::string fromPhoneme = fromPhonemeItem.key();
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if (!isSingleCodepoint(fromPhoneme)) {
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throw std::runtime_error(
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"Phonemes must be one codepoint (phoneme map)");
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}
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auto fromCodepoint = getCodepoint(fromPhoneme);
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for (auto &toPhonemeValue : fromPhonemeItem.value()) {
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std::string toPhoneme = toPhonemeValue.get<std::string>();
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if (!isSingleCodepoint(toPhoneme)) {
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throw std::runtime_error(
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"Phonemes must be one codepoint (phoneme map)");
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}
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auto toCodepoint = getCodepoint(toPhoneme);
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(*phonemizeConfig.phonemeMap)[fromCodepoint].push_back(toCodepoint);
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}
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}
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}
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// phoneme to [id] map
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if (configRoot.contains("phoneme_id_map")) {
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auto phonemeIdMapValue = configRoot["phoneme_id_map"];
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for (auto &fromPhonemeItem : phonemeIdMapValue.items()) {
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std::string fromPhoneme = fromPhonemeItem.key();
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if (!isSingleCodepoint(fromPhoneme)) {
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throw std::runtime_error(
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"Phonemes must be one codepoint (phoneme id map)");
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}
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auto fromCodepoint = getCodepoint(fromPhoneme);
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for (auto &toIdValue : fromPhonemeItem.value()) {
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PhonemeId toId = toIdValue.get<PhonemeId>();
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phonemizeConfig.phonemeIdMap[fromCodepoint].push_back(toId);
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}
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}
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}
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} /* parsePhonemizeConfig */
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void parseSynthesisConfig(json &configRoot, SynthesisConfig &synthesisConfig) {
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if (configRoot.contains("audio")) {
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auto audioValue = configRoot["audio"];
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if (audioValue.contains("sample_rate")) {
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// Default sample rate is 22050 Hz
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synthesisConfig.sampleRate = audioValue.value("sample_rate", 22050);
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}
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}
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} /* parseSynthesisConfig */
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void parseModelConfig(json &configRoot, ModelConfig &modelConfig) {
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modelConfig.numSpeakers = configRoot["num_speakers"].get<SpeakerId>();
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} /* parseModelConfig */
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void initialize(PiperConfig &config) {
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if (config.useESpeak) {
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// Set up espeak-ng for calling espeak_TextToPhonemesWithTerminator
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// See: https://github.com/rhasspy/espeak-ng
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int result = espeak_Initialize(AUDIO_OUTPUT_SYNCHRONOUS,
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/*buflength*/ 0,
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/*path*/ config.eSpeakDataPath.c_str(),
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/*options*/ 0);
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if (result < 0) {
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throw std::runtime_error("Failed to initialize eSpeak-ng");
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}
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}
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}
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|
||||
void terminate(PiperConfig &config) {
|
||||
if (config.useESpeak) {
|
||||
// Clean up espeak-ng
|
||||
espeak_Terminate();
|
||||
}
|
||||
}
|
||||
|
||||
void loadModel(std::string modelPath, ModelSession &session) {
|
||||
|
||||
session.env = Ort::Env(OrtLoggingLevel::ORT_LOGGING_LEVEL_WARNING,
|
||||
instanceName.c_str());
|
||||
session.env.DisableTelemetryEvents();
|
||||
|
||||
// Slows down performance by ~2x
|
||||
// session.options.SetIntraOpNumThreads(1);
|
||||
|
||||
// Roughly doubles load time for no visible inference benefit
|
||||
// session.options.SetGraphOptimizationLevel(
|
||||
// GraphOptimizationLevel::ORT_ENABLE_EXTENDED);
|
||||
|
||||
session.options.SetGraphOptimizationLevel(
|
||||
GraphOptimizationLevel::ORT_DISABLE_ALL);
|
||||
|
||||
// Slows down performance very slightly
|
||||
// session.options.SetExecutionMode(ExecutionMode::ORT_PARALLEL);
|
||||
|
||||
session.options.DisableCpuMemArena();
|
||||
session.options.DisableMemPattern();
|
||||
session.options.DisableProfiling();
|
||||
|
||||
auto startTime = std::chrono::steady_clock::now();
|
||||
session.onnx = Ort::Session(session.env, modelPath.c_str(), session.options);
|
||||
auto endTime = std::chrono::steady_clock::now();
|
||||
auto loadDuration = std::chrono::duration<double>(endTime - startTime);
|
||||
}
|
||||
|
||||
// Load Onnx model and JSON config file
|
||||
void loadVoice(PiperConfig &config, std::string modelPath,
|
||||
std::string modelConfigPath, Voice &voice,
|
||||
std::optional<SpeakerId> &speakerId) {
|
||||
std::ifstream modelConfigFile(modelConfigPath);
|
||||
voice.configRoot = json::parse(modelConfigFile);
|
||||
|
||||
parsePhonemizeConfig(voice.configRoot, voice.phonemizeConfig);
|
||||
parseSynthesisConfig(voice.configRoot, voice.synthesisConfig);
|
||||
parseModelConfig(voice.configRoot, voice.modelConfig);
|
||||
|
||||
if (voice.modelConfig.numSpeakers > 1) {
|
||||
// Multi-speaker model
|
||||
if (speakerId) {
|
||||
voice.synthesisConfig.speakerId = speakerId;
|
||||
} else {
|
||||
// Default speaker
|
||||
voice.synthesisConfig.speakerId = 0;
|
||||
}
|
||||
}
|
||||
|
||||
loadModel(modelPath, voice.session);
|
||||
|
||||
} /* loadVoice */
|
||||
|
||||
// Phoneme ids to WAV audio
|
||||
void synthesize(std::vector<PhonemeId> &phonemeIds,
|
||||
SynthesisConfig &synthesisConfig, ModelSession &session,
|
||||
std::vector<int16_t> &audioBuffer, SynthesisResult &result) {
|
||||
auto memoryInfo = Ort::MemoryInfo::CreateCpu(
|
||||
OrtAllocatorType::OrtArenaAllocator, OrtMemType::OrtMemTypeDefault);
|
||||
|
||||
// Allocate
|
||||
std::vector<int64_t> phonemeIdLengths{(int64_t)phonemeIds.size()};
|
||||
std::vector<float> scales{synthesisConfig.noiseScale,
|
||||
synthesisConfig.lengthScale,
|
||||
synthesisConfig.noiseW};
|
||||
|
||||
std::vector<Ort::Value> inputTensors;
|
||||
std::vector<int64_t> phonemeIdsShape{1, (int64_t)phonemeIds.size()};
|
||||
inputTensors.push_back(Ort::Value::CreateTensor<int64_t>(
|
||||
memoryInfo, phonemeIds.data(), phonemeIds.size(), phonemeIdsShape.data(),
|
||||
phonemeIdsShape.size()));
|
||||
|
||||
std::vector<int64_t> phomemeIdLengthsShape{(int64_t)phonemeIdLengths.size()};
|
||||
inputTensors.push_back(Ort::Value::CreateTensor<int64_t>(
|
||||
memoryInfo, phonemeIdLengths.data(), phonemeIdLengths.size(),
|
||||
phomemeIdLengthsShape.data(), phomemeIdLengthsShape.size()));
|
||||
|
||||
std::vector<int64_t> scalesShape{(int64_t)scales.size()};
|
||||
inputTensors.push_back(
|
||||
Ort::Value::CreateTensor<float>(memoryInfo, scales.data(), scales.size(),
|
||||
scalesShape.data(), scalesShape.size()));
|
||||
|
||||
// Add speaker id.
|
||||
// NOTE: These must be kept outside the "if" below to avoid being deallocated.
|
||||
std::vector<int64_t> speakerId{
|
||||
(int64_t)synthesisConfig.speakerId.value_or(0)};
|
||||
std::vector<int64_t> speakerIdShape{(int64_t)speakerId.size()};
|
||||
|
||||
if (synthesisConfig.speakerId) {
|
||||
inputTensors.push_back(Ort::Value::CreateTensor<int64_t>(
|
||||
memoryInfo, speakerId.data(), speakerId.size(), speakerIdShape.data(),
|
||||
speakerIdShape.size()));
|
||||
}
|
||||
|
||||
// From export_onnx.py
|
||||
std::array<const char *, 4> inputNames = {"input", "input_lengths", "scales",
|
||||
"sid"};
|
||||
std::array<const char *, 1> outputNames = {"output"};
|
||||
|
||||
// Infer
|
||||
auto startTime = std::chrono::steady_clock::now();
|
||||
auto outputTensors = session.onnx.Run(
|
||||
Ort::RunOptions{nullptr}, inputNames.data(), inputTensors.data(),
|
||||
inputTensors.size(), outputNames.data(), outputNames.size());
|
||||
auto endTime = std::chrono::steady_clock::now();
|
||||
|
||||
if ((outputTensors.size() != 1) || (!outputTensors.front().IsTensor())) {
|
||||
throw std::runtime_error("Invalid output tensors");
|
||||
}
|
||||
auto inferDuration = std::chrono::duration<double>(endTime - startTime);
|
||||
result.inferSeconds = inferDuration.count();
|
||||
|
||||
const float *audio = outputTensors.front().GetTensorData<float>();
|
||||
auto audioShape =
|
||||
outputTensors.front().GetTensorTypeAndShapeInfo().GetShape();
|
||||
int64_t audioCount = audioShape[audioShape.size() - 1];
|
||||
|
||||
result.audioSeconds = (double)audioCount / (double)synthesisConfig.sampleRate;
|
||||
result.realTimeFactor = 0.0;
|
||||
if (result.audioSeconds > 0) {
|
||||
result.realTimeFactor = result.inferSeconds / result.audioSeconds;
|
||||
}
|
||||
|
||||
// Get max audio value for scaling
|
||||
float maxAudioValue = 0.01f;
|
||||
for (int64_t i = 0; i < audioCount; i++) {
|
||||
float audioValue = abs(audio[i]);
|
||||
if (audioValue > maxAudioValue) {
|
||||
maxAudioValue = audioValue;
|
||||
}
|
||||
}
|
||||
|
||||
// We know the size up front
|
||||
audioBuffer.reserve(audioCount);
|
||||
|
||||
// Scale audio to fill range and convert to int16
|
||||
float audioScale = (MAX_WAV_VALUE / std::max(0.01f, maxAudioValue));
|
||||
for (int64_t i = 0; i < audioCount; i++) {
|
||||
int16_t intAudioValue = static_cast<int16_t>(
|
||||
std::clamp(audio[i] * audioScale,
|
||||
static_cast<float>(std::numeric_limits<int16_t>::min()),
|
||||
static_cast<float>(std::numeric_limits<int16_t>::max())));
|
||||
|
||||
audioBuffer.push_back(intAudioValue);
|
||||
}
|
||||
|
||||
// Clean up
|
||||
for (std::size_t i = 0; i < outputTensors.size(); i++) {
|
||||
Ort::detail::OrtRelease(outputTensors[i].release());
|
||||
}
|
||||
|
||||
for (std::size_t i = 0; i < inputTensors.size(); i++) {
|
||||
Ort::detail::OrtRelease(inputTensors[i].release());
|
||||
}
|
||||
}
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
// Phonemize text and synthesize audio
|
||||
void textToAudio(PiperConfig &config, Voice &voice, std::string text,
|
||||
std::vector<int16_t> &audioBuffer, SynthesisResult &result,
|
||||
const std::function<void()> &audioCallback) {
|
||||
|
||||
std::size_t sentenceSilenceSamples = 0;
|
||||
if (voice.synthesisConfig.sentenceSilenceSeconds > 0) {
|
||||
sentenceSilenceSamples = (std::size_t)(
|
||||
voice.synthesisConfig.sentenceSilenceSeconds *
|
||||
voice.synthesisConfig.sampleRate * voice.synthesisConfig.channels);
|
||||
}
|
||||
|
||||
// Phonemes for each sentence
|
||||
std::vector<std::vector<Phoneme>> phonemes;
|
||||
|
||||
if (voice.phonemizeConfig.phonemeType == eSpeakPhonemes) {
|
||||
// Use espeak-ng for phonemization
|
||||
eSpeakPhonemeConfig eSpeakConfig;
|
||||
eSpeakConfig.voice = voice.phonemizeConfig.eSpeak->voice;
|
||||
phonemize_eSpeak(text, eSpeakConfig, phonemes);
|
||||
} else {
|
||||
// Use UTF-8 codepoints as "phonemes"
|
||||
CodepointsPhonemeConfig codepointsConfig;
|
||||
phonemize_codepoints(text, codepointsConfig, phonemes);
|
||||
}
|
||||
|
||||
// Synthesize each sentence independently.
|
||||
std::vector<PhonemeId> phonemeIds;
|
||||
std::map<Phoneme, std::size_t> missingPhonemes;
|
||||
for (auto phonemesIter = phonemes.begin(); phonemesIter != phonemes.end();
|
||||
++phonemesIter) {
|
||||
std::vector<Phoneme> &sentencePhonemes = *phonemesIter;
|
||||
SynthesisResult sentenceResult;
|
||||
|
||||
PhonemeIdConfig idConfig;
|
||||
if (voice.phonemizeConfig.phonemeType == TextPhonemes) {
|
||||
auto &language = voice.phonemizeConfig.eSpeak->voice;
|
||||
if (DEFAULT_ALPHABET.count(language) < 1) {
|
||||
throw std::runtime_error(
|
||||
"Text phoneme language for voice is not supported");
|
||||
}
|
||||
|
||||
// Use alphabet for language
|
||||
idConfig.phonemeIdMap =
|
||||
std::make_shared<PhonemeIdMap>(DEFAULT_ALPHABET[language]);
|
||||
}
|
||||
|
||||
// phonemes -> ids
|
||||
phonemes_to_ids(sentencePhonemes, idConfig, phonemeIds, missingPhonemes);
|
||||
|
||||
// ids -> audio
|
||||
synthesize(phonemeIds, voice.synthesisConfig, voice.session, audioBuffer,
|
||||
sentenceResult);
|
||||
|
||||
// Add end of sentence silence
|
||||
if (sentenceSilenceSamples > 0) {
|
||||
for (std::size_t i = 0; i < sentenceSilenceSamples; i++) {
|
||||
audioBuffer.push_back(0);
|
||||
}
|
||||
}
|
||||
|
||||
if (audioCallback) {
|
||||
// Call back must copy audio since it is cleared afterwards.
|
||||
audioCallback();
|
||||
audioBuffer.clear();
|
||||
}
|
||||
|
||||
result.audioSeconds += sentenceResult.audioSeconds;
|
||||
result.inferSeconds += sentenceResult.inferSeconds;
|
||||
|
||||
phonemeIds.clear();
|
||||
}
|
||||
|
||||
if (result.audioSeconds > 0) {
|
||||
result.realTimeFactor = result.inferSeconds / result.audioSeconds;
|
||||
}
|
||||
|
||||
} /* textToAudio */
|
||||
|
||||
// Phonemize text and synthesize audio to WAV file
|
||||
void textToWavFile(PiperConfig &config, Voice &voice, std::string text,
|
||||
std::ostream &audioFile, SynthesisResult &result) {
|
||||
|
||||
std::vector<int16_t> audioBuffer;
|
||||
textToAudio(config, voice, text, audioBuffer, result, NULL);
|
||||
|
||||
// Write WAV
|
||||
auto synthesisConfig = voice.synthesisConfig;
|
||||
writeWavHeader(synthesisConfig.sampleRate, synthesisConfig.sampleWidth,
|
||||
synthesisConfig.channels, (int32_t)audioBuffer.size(),
|
||||
audioFile);
|
||||
|
||||
audioFile.write((const char *)audioBuffer.data(),
|
||||
sizeof(int16_t) * audioBuffer.size());
|
||||
|
||||
} /* textToWavFile */
|
||||
|
||||
} // namespace piper
|
||||
@ -1,130 +0,0 @@
|
||||
#ifndef SYNTHESIZE_H_
|
||||
#define SYNTHESIZE_H_
|
||||
|
||||
#include <array>
|
||||
#include <chrono>
|
||||
#include <limits>
|
||||
#include <memory>
|
||||
#include <vector>
|
||||
|
||||
#include <onnxruntime_cxx_api.h>
|
||||
|
||||
#include "config.hpp"
|
||||
#include "model.hpp"
|
||||
|
||||
using namespace std;
|
||||
|
||||
namespace piper {
|
||||
|
||||
// Maximum value for 16-bit signed WAV sample
|
||||
const float MAX_WAV_VALUE = 32767.0f;
|
||||
|
||||
struct SynthesisResult {
|
||||
double inferSeconds;
|
||||
double audioSeconds;
|
||||
double realTimeFactor;
|
||||
};
|
||||
|
||||
// Phoneme ids to WAV audio
|
||||
void synthesize(vector<PhonemeId> &phonemeIds, SynthesisConfig &synthesisConfig,
|
||||
ModelSession &session, vector<int16_t> &audioBuffer,
|
||||
SynthesisResult &result) {
|
||||
auto memoryInfo = Ort::MemoryInfo::CreateCpu(
|
||||
OrtAllocatorType::OrtArenaAllocator, OrtMemType::OrtMemTypeDefault);
|
||||
|
||||
// Allocate
|
||||
vector<int64_t> phonemeIdLengths{(int64_t)phonemeIds.size()};
|
||||
vector<float> scales{synthesisConfig.noiseScale, synthesisConfig.lengthScale,
|
||||
synthesisConfig.noiseW};
|
||||
|
||||
vector<Ort::Value> inputTensors;
|
||||
vector<int64_t> phonemeIdsShape{1, (int64_t)phonemeIds.size()};
|
||||
inputTensors.push_back(Ort::Value::CreateTensor<int64_t>(
|
||||
memoryInfo, phonemeIds.data(), phonemeIds.size(), phonemeIdsShape.data(),
|
||||
phonemeIdsShape.size()));
|
||||
|
||||
vector<int64_t> phomemeIdLengthsShape{(int64_t)phonemeIdLengths.size()};
|
||||
inputTensors.push_back(Ort::Value::CreateTensor<int64_t>(
|
||||
memoryInfo, phonemeIdLengths.data(), phonemeIdLengths.size(),
|
||||
phomemeIdLengthsShape.data(), phomemeIdLengthsShape.size()));
|
||||
|
||||
vector<int64_t> scalesShape{(int64_t)scales.size()};
|
||||
inputTensors.push_back(
|
||||
Ort::Value::CreateTensor<float>(memoryInfo, scales.data(), scales.size(),
|
||||
scalesShape.data(), scalesShape.size()));
|
||||
|
||||
// Add speaker id.
|
||||
// NOTE: These must be kept outside the "if" below to avoid being deallocated.
|
||||
vector<int64_t> speakerId{(int64_t)synthesisConfig.speakerId.value_or(0)};
|
||||
vector<int64_t> speakerIdShape{(int64_t)speakerId.size()};
|
||||
|
||||
if (synthesisConfig.speakerId) {
|
||||
inputTensors.push_back(Ort::Value::CreateTensor<int64_t>(
|
||||
memoryInfo, speakerId.data(), speakerId.size(), speakerIdShape.data(),
|
||||
speakerIdShape.size()));
|
||||
}
|
||||
|
||||
// From export_onnx.py
|
||||
array<const char *, 4> inputNames = {"input", "input_lengths", "scales",
|
||||
"sid"};
|
||||
array<const char *, 1> outputNames = {"output"};
|
||||
|
||||
// Infer
|
||||
auto startTime = chrono::steady_clock::now();
|
||||
auto outputTensors = session.onnx.Run(
|
||||
Ort::RunOptions{nullptr}, inputNames.data(), inputTensors.data(),
|
||||
inputTensors.size(), outputNames.data(), outputNames.size());
|
||||
auto endTime = chrono::steady_clock::now();
|
||||
|
||||
if ((outputTensors.size() != 1) || (!outputTensors.front().IsTensor())) {
|
||||
throw runtime_error("Invalid output tensors");
|
||||
}
|
||||
auto inferDuration = chrono::duration<double>(endTime - startTime);
|
||||
result.inferSeconds = inferDuration.count();
|
||||
|
||||
const float *audio = outputTensors.front().GetTensorData<float>();
|
||||
auto audioShape =
|
||||
outputTensors.front().GetTensorTypeAndShapeInfo().GetShape();
|
||||
int64_t audioCount = audioShape[audioShape.size() - 1];
|
||||
|
||||
result.audioSeconds = (double)audioCount / (double)synthesisConfig.sampleRate;
|
||||
result.realTimeFactor = 0.0;
|
||||
if (result.audioSeconds > 0) {
|
||||
result.realTimeFactor = result.inferSeconds / result.audioSeconds;
|
||||
}
|
||||
|
||||
// Get max audio value for scaling
|
||||
float maxAudioValue = 0.01f;
|
||||
for (int64_t i = 0; i < audioCount; i++) {
|
||||
float audioValue = abs(audio[i]);
|
||||
if (audioValue > maxAudioValue) {
|
||||
maxAudioValue = audioValue;
|
||||
}
|
||||
}
|
||||
|
||||
// We know the size up front
|
||||
audioBuffer.reserve(audioCount);
|
||||
|
||||
// Scale audio to fill range and convert to int16
|
||||
float audioScale = (MAX_WAV_VALUE / max(0.01f, maxAudioValue));
|
||||
for (int64_t i = 0; i < audioCount; i++) {
|
||||
int16_t intAudioValue = static_cast<int16_t>(
|
||||
clamp(audio[i] * audioScale,
|
||||
static_cast<float>(numeric_limits<int16_t>::min()),
|
||||
static_cast<float>(numeric_limits<int16_t>::max())));
|
||||
|
||||
audioBuffer.push_back(intAudioValue);
|
||||
}
|
||||
|
||||
// Clean up
|
||||
for (size_t i = 0; i < outputTensors.size(); i++) {
|
||||
Ort::detail::OrtRelease(outputTensors[i].release());
|
||||
}
|
||||
|
||||
for (size_t i = 0; i < inputTensors.size(); i++) {
|
||||
Ort::detail::OrtRelease(inputTensors[i].release());
|
||||
}
|
||||
}
|
||||
} // namespace piper
|
||||
|
||||
#endif // SYNTHESIZE_H_
|
||||
Loading…
Reference in New Issue