diff --git a/CppDataProcess/F0Preprocess.cpp b/CppDataProcess/F0Preprocess.cpp new file mode 100644 index 0000000000000000000000000000000000000000..d6bd6f3cb8033fb9263624a9707311cac593ad57 --- /dev/null +++ b/CppDataProcess/F0Preprocess.cpp @@ -0,0 +1,153 @@ +#include "F0Preprocess.hpp" + + +void F0PreProcess::compute_f0(const double* audio, int64_t len) +{ + DioOption Doption; + InitializeDioOption(&Doption); + Doption.f0_ceil = 800; + Doption.frame_period = 1000.0 * hop / fs; + f0Len = GetSamplesForDIO(fs, (int)len, Doption.frame_period); + const auto tp = new double[f0Len]; + const auto tmpf0 = new double[f0Len]; + rf0 = new double[f0Len]; + Dio(audio, (int)len, fs, &Doption, tp, tmpf0); + StoneMask(audio, (int)len, fs, tp, tmpf0, (int)f0Len, rf0); + delete[] tmpf0; + delete[] tp; +} + +std::vector arange(double start,double end,double step = 1.0,double div = 1.0) +{ + std::vector output; + while(start(f0Len), xi.data(), (int)xi.size(), tmp); + for (size_t i = 0; i < xi.size(); i++) + if (isnan(tmp[i])) + tmp[i] = 0.0; + delete[] rf0; + rf0 = nullptr; + rf0 = tmp; + f0Len = (int64_t)xi.size(); +} + +long long* F0PreProcess::f0Log() +{ + const auto tmp = new long long[f0Len]; + const auto f0_mel = new double[f0Len]; + for (long long i = 0; i < f0Len; i++) + { + f0_mel[i] = 1127 * log(1.0 + rf0[i] / 700.0); + if (f0_mel[i] > 0.0) + f0_mel[i] = (f0_mel[i] - f0_mel_min) * (f0_bin - 2.0) / (f0_mel_max - f0_mel_min) + 1.0; + if (f0_mel[i] < 1.0) + f0_mel[i] = 1; + if (f0_mel[i] > f0_bin - 1) + f0_mel[i] = f0_bin - 1; + tmp[i] = (long long)round(f0_mel[i]); + } + delete[] f0_mel; + delete[] rf0; + rf0 = nullptr; + return tmp; +} + +std::vector F0PreProcess::GetF0AndOtherInput(const double* audio, int64_t audioLen, int64_t hubLen, int64_t tran) +{ + compute_f0(audio, audioLen); + for (int64_t i = 0; i < f0Len; ++i) + { + rf0[i] = rf0[i] * pow(2.0, static_cast(tran) / 12.0); + if (rf0[i] < 0.001) + rf0[i] = NAN; + } + InterPf0(hubLen); + const auto O0f = f0Log(); + std::vector Of0(O0f, O0f + f0Len); + delete[] O0f; + return Of0; +} + +std::vector getAligments(size_t specLen, size_t hubertLen) +{ + std::vector mel2ph(specLen + 1, 0); + + size_t startFrame = 0; + const double ph_durs = static_cast(specLen) / static_cast(hubertLen); + for (size_t iph = 0; iph < hubertLen; ++iph) + { + const auto endFrame = static_cast(round(static_cast(iph) * ph_durs + ph_durs)); + for (auto j = startFrame; j < endFrame + 1; ++j) + mel2ph[j] = static_cast(iph) + 1; + startFrame = endFrame + 1; + } + + return mel2ph; +} + +std::vector F0PreProcess::GetF0AndOtherInputF0(const double* audio, int64_t audioLen, int64_t tran) +{ + compute_f0(audio, audioLen); + for (int64_t i = 0; i < f0Len; ++i) + { + rf0[i] = log2(rf0[i] * pow(2.0, static_cast(tran) / 12.0)); + if (rf0[i] < 0.001) + rf0[i] = NAN; + } + const int64_t specLen = audioLen / hop; + InterPf0(specLen); + + std::vector Of0(specLen, 0.0); + + double last_value = 0.0; + for (int64_t i = 0; i < specLen; ++i) + { + if (rf0[i] <= 0.0) + { + int64_t j = i + 1; + for (; j < specLen; ++j) + { + if (rf0[j] > 0.0) + break; + } + if (j < specLen - 1) + { + if (last_value > 0.0) + { + const auto step = (rf0[j] - rf0[i - 1]) / double(j - i); + for (int64_t k = i; k < j; ++k) + Of0[k] = float(rf0[i - 1] + step * double(k - i + 1)); + } + else + for (int64_t k = i; k < j; ++k) + Of0[k] = float(rf0[j]); + i = j; + } + else + { + for (int64_t k = i; k < specLen; ++k) + Of0[k] = float(last_value); + i = specLen; + } + } + else + { + Of0[i] = float(rf0[i - 1]); + last_value = rf0[i]; + } + } + delete[] rf0; + rf0 = nullptr; + return Of0; +} diff --git a/CppDataProcess/F0Preprocess.hpp b/CppDataProcess/F0Preprocess.hpp new file mode 100644 index 0000000000000000000000000000000000000000..7816bb5a7d7c4d805f71fc8dd3b128e5eb2deb47 --- /dev/null +++ b/CppDataProcess/F0Preprocess.hpp @@ -0,0 +1,36 @@ +#include "world/dio.h" +#include "world/stonemask.h" +#include "world/matlabfunctions.h" +#include +#include + +//Cpp F0 Preprocess + +class F0PreProcess +{ +public: + int fs; + short hop; + const int f0_bin = 256; + const double f0_max = 1100.0; + const double f0_min = 50.0; + const double f0_mel_min = 1127.0 * log(1.0 + f0_min / 700.0); + const double f0_mel_max = 1127.0 * log(1.0 + f0_max / 700.0); + F0PreProcess(int sr = 16000, short h = 160) :fs(sr), hop(h) {} + ~F0PreProcess() + { + delete[] rf0; + rf0 = nullptr; + } + void compute_f0(const double* audio, int64_t len); + void InterPf0(int64_t len); + long long* f0Log(); + int64_t getLen()const { return f0Len; } + std::vector GetF0AndOtherInput(const double* audio, int64_t audioLen, int64_t hubLen, int64_t tran); + std::vector GetF0AndOtherInputF0(const double* audio, int64_t audioLen, int64_t tran); +private: + double* rf0 = nullptr; + int64_t f0Len = 0; +}; + +std::vector getAligments(size_t specLen, size_t hubertLen); diff --git a/CppDataProcess/Slicer.hpp b/CppDataProcess/Slicer.hpp new file mode 100644 index 0000000000000000000000000000000000000000..dfea70caaedce33dcb08fb21c0010575387c36f0 --- /dev/null +++ b/CppDataProcess/Slicer.hpp @@ -0,0 +1,82 @@ +#include +#include +#include "Wav.hpp" + +struct SliceResult +{ + std::vector SliceOffset; + std::vector SliceTag; + cutResult(std::vector&& O, std::vector&& T) :SliceOffset(O), SliceTag(T) {} +}; + +double getAvg(const short* start, const short* end) +{ + const auto size = end - start + 1; + auto avg = (double)(*start); + for (auto i = 1; i < size; i++) + { + avg = avg + (abs((double)start[i]) - avg) / (double)(i + 1ull); + } + return avg; +} + +inline SliceResult SliceWav(Wav& input, double threshold, unsigned long minLen, unsigned short frame_len, unsigned short frame_shift) +{ + const auto header = input.getHeader(); + if (header.Subchunk2Size < minLen * header.bytesPerSec) + return { {0,header.Subchunk2Size},{true} }; + auto ptr = input.getData(); + std::vector output; + std::vector tag; + auto n = (header.Subchunk2Size / frame_shift) - 2 * (frame_len / frame_shift); + unsigned long nn = 0; + bool cutTag = true; + output.emplace_back(0); + while (n--) + { + //if (nn > minLen * header.bytesPerSec) + if (cutTag) + { + const auto vol = abs(getAvg((short*)ptr, (short*)ptr + frame_len)); + if (vol < threshold) + { + cutTag = false; + if (nn > minLen * header.bytesPerSec) + { + nn = 0; + output.emplace_back((ptr - input.getData()) + (frame_len / 2)); + } + } + else + { + cutTag = true; + } + } + else + { + const auto vol = abs(getAvg((short*)ptr, (short*)ptr + frame_len)); + if (vol < threshold) + { + cutTag = false; + } + else + { + cutTag = true; + if (nn > minLen * header.bytesPerSec) + { + nn = 0; + output.emplace_back((ptr - input.getData()) + (frame_len / 2)); + } + } + } + nn += frame_shift; + ptr += frame_shift; + } + output.push_back(header.Subchunk2Size); + for (size_t i = 1; i < output.size(); i++) + { + tag.push_back(abs(getAvg((short*)(input.getData() + output[i - 1]), (short*)(input.getData() + output[i]))) > threshold); + } + return { std::move(output),std::move(tag) }; +} + diff --git a/CppDataProcess/Wav.cpp b/CppDataProcess/Wav.cpp new file mode 100644 index 0000000000000000000000000000000000000000..c2b7e8a9421d179e6001e8a9483d0e427833e952 --- /dev/null +++ b/CppDataProcess/Wav.cpp @@ -0,0 +1,151 @@ +#include "Wav.hpp" + +Wav::Wav(const wchar_t* Path) :header(WAV_HEADER()) { + char buf[1024]; + FILE* stream; + _wfreopen_s(&stream, Path, L"rb", stderr); + if (stream == nullptr) { + throw (std::exception("File not exists")); + } + fread(buf, 1, HEAD_LENGTH, stream); + int pos = 0; + while (pos < HEAD_LENGTH) { + if ((buf[pos] == 'R') && (buf[pos + 1] == 'I') && (buf[pos + 2] == 'F') && (buf[pos + 3] == 'F')) { + pos += 4; + break; + } + ++pos; + } + if (pos >= HEAD_LENGTH) + throw (std::exception("Don't order fried rice (annoyed)")); + header.ChunkSize = *(int*)&buf[pos]; + pos += 8; + while (pos < HEAD_LENGTH) { + if ((buf[pos] == 'f') && (buf[pos + 1] == 'm') && (buf[pos + 2] == 't')) { + pos += 4; + break; + } + ++pos; + } + if (pos >= HEAD_LENGTH) + throw (std::exception("Don't order fried rice (annoyed)")); + header.Subchunk1Size = *(int*)&buf[pos]; + pos += 4; + header.AudioFormat = *(short*)&buf[pos]; + pos += 2; + header.NumOfChan = *(short*)&buf[pos]; + pos += 2; + header.SamplesPerSec = *(int*)&buf[pos]; + pos += 4; + header.bytesPerSec = *(int*)&buf[pos]; + pos += 4; + header.blockAlign = *(short*)&buf[pos]; + pos += 2; + header.bitsPerSample = *(short*)&buf[pos]; + pos += 2; + while (pos < HEAD_LENGTH) { + if ((buf[pos] == 'd') && (buf[pos + 1] == 'a') && (buf[pos + 2] == 't') && (buf[pos + 3] == 'a')) { + pos += 4; + break; + } + ++pos; + } + if (pos >= HEAD_LENGTH) + throw (std::exception("Don't order fried rice (annoyed)")); + header.Subchunk2Size = *(int*)&buf[pos]; + pos += 4; + StartPos = pos; + Data = new char[header.Subchunk2Size + 1]; + fseek(stream, StartPos, SEEK_SET); + fread(Data, 1, header.Subchunk2Size, stream); + if (stream != nullptr) { + fclose(stream); + } + SData = reinterpret_cast(Data); + dataSize = header.Subchunk2Size / 2; +} + +Wav::Wav(const Wav& input) :header(WAV_HEADER()) { + Data = new char[(input.header.Subchunk2Size + 1)]; + if (Data == nullptr) { throw std::exception("OOM"); } + memcpy(header.RIFF, input.header.RIFF, 4); + memcpy(header.fmt, input.header.fmt, 4); + memcpy(header.WAVE, input.header.WAVE, 4); + memcpy(header.Subchunk2ID, input.header.Subchunk2ID, 4); + header.ChunkSize = input.header.ChunkSize; + header.Subchunk1Size = input.header.Subchunk1Size; + header.AudioFormat = input.header.AudioFormat; + header.NumOfChan = input.header.NumOfChan; + header.SamplesPerSec = input.header.SamplesPerSec; + header.bytesPerSec = input.header.bytesPerSec; + header.blockAlign = input.header.blockAlign; + header.bitsPerSample = input.header.bitsPerSample; + header.Subchunk2Size = input.header.Subchunk2Size; + StartPos = input.StartPos; + memcpy(Data, input.Data, input.header.Subchunk2Size); + SData = reinterpret_cast(Data); + dataSize = header.Subchunk2Size / 2; +} + +Wav::Wav(Wav&& input) noexcept +{ + Data = input.Data; + input.Data = nullptr; + memcpy(header.RIFF, input.header.RIFF, 4); + memcpy(header.fmt, input.header.fmt, 4); + memcpy(header.WAVE, input.header.WAVE, 4); + memcpy(header.Subchunk2ID, input.header.Subchunk2ID, 4); + header.ChunkSize = input.header.ChunkSize; + header.Subchunk1Size = input.header.Subchunk1Size; + header.AudioFormat = input.header.AudioFormat; + header.NumOfChan = input.header.NumOfChan; + header.SamplesPerSec = input.header.SamplesPerSec; + header.bytesPerSec = input.header.bytesPerSec; + header.blockAlign = input.header.blockAlign; + header.bitsPerSample = input.header.bitsPerSample; + header.Subchunk2Size = input.header.Subchunk2Size; + StartPos = input.StartPos; + SData = reinterpret_cast(Data); + dataSize = header.Subchunk2Size / 2; +} + +Wav& Wav::operator=(Wav&& input) noexcept +{ + destory(); + Data = input.Data; + input.Data = nullptr; + memcpy(header.RIFF, input.header.RIFF, 4); + memcpy(header.fmt, input.header.fmt, 4); + memcpy(header.WAVE, input.header.WAVE, 4); + memcpy(header.Subchunk2ID, input.header.Subchunk2ID, 4); + header.ChunkSize = input.header.ChunkSize; + header.Subchunk1Size = input.header.Subchunk1Size; + header.AudioFormat = input.header.AudioFormat; + header.NumOfChan = input.header.NumOfChan; + header.SamplesPerSec = input.header.SamplesPerSec; + header.bytesPerSec = input.header.bytesPerSec; + header.blockAlign = input.header.blockAlign; + header.bitsPerSample = input.header.bitsPerSample; + header.Subchunk2Size = input.header.Subchunk2Size; + StartPos = input.StartPos; + SData = reinterpret_cast(Data); + dataSize = header.Subchunk2Size / 2; + return *this; +} + +Wav& Wav::cat(const Wav& input) +{ + if (header.AudioFormat != 1) return *this; + if (header.SamplesPerSec != input.header.bitsPerSample || header.NumOfChan != input.header.NumOfChan) return *this; + char* buffer = new char[(int64_t)header.Subchunk2Size + (int64_t)input.header.Subchunk2Size + 1]; + if (buffer == nullptr)return *this; + memcpy(buffer, Data, header.Subchunk2Size); + memcpy(buffer + header.Subchunk2Size, input.Data, input.header.Subchunk2Size); + header.ChunkSize += input.header.Subchunk2Size; + header.Subchunk2Size += input.header.Subchunk2Size; + delete[] Data; + Data = buffer; + SData = reinterpret_cast(Data); + dataSize = header.Subchunk2Size / 2; + return *this; +} diff --git a/CppDataProcess/Wav.hpp b/CppDataProcess/Wav.hpp new file mode 100644 index 0000000000000000000000000000000000000000..c633366256a47fad29f8a385e03f847c0c94d1cb --- /dev/null +++ b/CppDataProcess/Wav.hpp @@ -0,0 +1,99 @@ +class Wav { +public: + + struct WAV_HEADER { + char RIFF[4] = { 'R','I','F','F' }; //RIFF��ʶ + unsigned long ChunkSize; //�ļ���С-8 + char WAVE[4] = { 'W','A','V','E' }; //WAVE�� + char fmt[4] = { 'f','m','t',' ' }; //fmt�� + unsigned long Subchunk1Size; //fmt���С + unsigned short AudioFormat; //�����ʽ + unsigned short NumOfChan; //������ + unsigned long SamplesPerSec; //������ + unsigned long bytesPerSec; //ÿ�����ֽ��� + unsigned short blockAlign; //�������ֽ� + unsigned short bitsPerSample; //������λ�� + char Subchunk2ID[4] = { 'd','a','t','a' }; //���ݿ� + unsigned long Subchunk2Size; //���ݿ��С + WAV_HEADER(unsigned long cs = 36, unsigned long sc1s = 16, unsigned short af = 1, unsigned short nc = 1, unsigned long sr = 22050, unsigned long bps = 44100, unsigned short ba = 2, unsigned short bips = 16, unsigned long sc2s = 0) :ChunkSize(cs), Subchunk1Size(sc1s), AudioFormat(af), NumOfChan(nc), SamplesPerSec(sr), bytesPerSec(bps), blockAlign(ba), bitsPerSample(bips), Subchunk2Size(sc2s) {} + }; + using iterator = int16_t*; + Wav(unsigned long cs = 36, unsigned long sc1s = 16, unsigned short af = 1, unsigned short nc = 1, unsigned long sr = 22050, unsigned long bps = 44100, unsigned short ba = 2, unsigned short bips = 16, unsigned long sc2s = 0) :header({ + cs, + sc1s, + af, + nc, + sr, + bps, + ba, + bips, + sc2s + }), Data(nullptr), StartPos(44) { + dataSize = 0; + SData = nullptr; + } + Wav(unsigned long sr, unsigned long length, const void* data) :header({ + 36, + 16, + 1, + 1, + sr, + sr * 2, + 2, + 16, + length + }), Data(new char[length + 1]), StartPos(44) + { + header.ChunkSize = 36 + length; + memcpy(Data, data, length); + SData = reinterpret_cast(Data); + dataSize = length / 2; + } + Wav(const wchar_t* Path); + Wav(const Wav& input); + Wav(Wav&& input) noexcept; + Wav& operator=(const Wav& input) = delete; + Wav& operator=(Wav&& input) noexcept; + ~Wav() { destory(); } + Wav& cat(const Wav& input); + bool isEmpty() const { return this->header.Subchunk2Size == 0; } + const char* getData() const { return Data; } + char* getData() { return Data; } + WAV_HEADER getHeader() const { return header; } + WAV_HEADER& Header() { return header; } + void destory() const { delete[] Data; } + void changeData(const void* indata,long length,int sr) + { + delete[] Data; + Data = new char[length]; + memcpy(Data, indata, length); + header.ChunkSize = 36 + length; + header.Subchunk2Size = length; + header.SamplesPerSec = sr; + header.bytesPerSec = 2 * sr; + } + int16_t& operator[](const size_t index) const + { + if (index < dataSize) + return *(SData + index); + return *(SData + dataSize - 1); + } + iterator begin() const + { + return reinterpret_cast(Data); + } + iterator end() const + { + return reinterpret_cast(Data + header.Subchunk2Size); + } + int64_t getDataLen()const + { + return static_cast(dataSize); + } +private: + WAV_HEADER header; + char* Data; + int16_t* SData; + size_t dataSize; + int StartPos; +}; diff --git a/CppDataProcess/readme.md b/CppDataProcess/readme.md new file mode 100644 index 0000000000000000000000000000000000000000..ca6f7d0d011eeefd7f18ff601cff5507c80b3fc7 --- /dev/null +++ b/CppDataProcess/readme.md @@ -0,0 +1,8 @@ +## F0Preprocess +请前往 https://github.com/JeremyCCHsu/Python-Wrapper-for-World-Vocoder 下载PyWorld的源代码并编译出静态库并链接到你的项目之中，然后调用此头文件 + +## Slicer +一个简单的切片机 + +--- +~~上面的东西是直接从MoeSS的代码里面抽出来的，可以作为预置预处理的替代品（）~~ diff --git a/LICENSE b/LICENSE new file mode 100644 index 0000000000000000000000000000000000000000..c7202d4281303c431d24ad9a0e3a24a0b37517f3 --- /dev/null +++ b/LICENSE @@ -0,0 +1,21 @@ +MIT License + +Copyright (c) 2021 Jingyi Li + +Permission is hereby granted, free of charge, to any person obtaining a copy +of this software and associated documentation files (the "Software"), to deal +in the Software without restriction, including without limitation the rights +to use, copy, modify, merge, publish, distribute, sublicense, and/or sell +copies of the Software, and to permit persons to whom the Software is +furnished to do so, subject to the following conditions: + +The above copyright notice and this permission notice shall be included in all +copies or substantial portions of the Software. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. diff --git a/app.py b/app.py new file mode 100644 index 0000000000000000000000000000000000000000..0ff0c88cb2348521aa0a84b86d49002343de20f5 --- /dev/null +++ b/app.py @@ -0,0 +1,69 @@ +import io +import os + +# os.system("wget -P cvec/ https://huggingface.co/spaces/innnky/nanami/resolve/main/checkpoint_best_legacy_500.pt") +import gradio as gr +import librosa +import numpy as np +import soundfile +from inference.infer_tool import Svc +import logging + +logging.getLogger('numba').setLevel(logging.WARNING) +logging.getLogger('markdown_it').setLevel(logging.WARNING) +logging.getLogger('urllib3').setLevel(logging.WARNING) +logging.getLogger('matplotlib').setLevel(logging.WARNING) + +config_path = "configs/config.json" + +model = Svc("logs/44k/G_114400.pth", "configs/config.json", cluster_model_path="logs/44k/kmeans_10000.pt") + + + +def vc_fn(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale): + if input_audio is None: + return "You need to upload an audio", None + sampling_rate, audio = input_audio + # print(audio.shape,sampling_rate) + duration = audio.shape[0] / sampling_rate + if duration > 90: + return "请上传小于90s的音频，需要转换长音频请本地进行转换", None + audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32) + if len(audio.shape) > 1: + audio = librosa.to_mono(audio.transpose(1, 0)) + if sampling_rate != 16000: + audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000) + print(audio.shape) + out_wav_path = "temp.wav" + soundfile.write(out_wav_path, audio, 16000, format="wav") + print( cluster_ratio, auto_f0, noise_scale) + _audio = model.slice_inference(out_wav_path, sid, vc_transform, slice_db, cluster_ratio, auto_f0, noise_scale) + return "Success", (44100, _audio) + + +app = gr.Blocks() +with app: + with gr.Tabs(): + with gr.TabItem("Basic"): + gr.Markdown(value=""" + sovits4.0 在线demo + + 此demo为预训练底模在线demo，使用数据：云灏 即霜 辉宇·星AI 派蒙 绫地宁宁 + """) + spks = list(model.spk2id.keys()) + sid = gr.Dropdown(label="音色", choices=spks, value=spks[0]) + vc_input3 = gr.Audio(label="上传音频（长度小于90秒）") + vc_transform = gr.Number(label="变调（整数，可以正负，半音数量，升高八度就是12）", value=0) + cluster_ratio = gr.Number(label="聚类模型混合比例，0-1之间，默认为0不启用聚类，能提升音色相似度，但会导致咬字下降（如果使用建议0.5左右）", value=0) + auto_f0 = gr.Checkbox(label="自动f0预测，配合聚类模型f0预测效果更好,会导致变调功能失效（仅限转换语音，歌声不要勾选此项会究极跑调）", value=False) + slice_db = gr.Number(label="切片阈值", value=-40) + noise_scale = gr.Number(label="noise_scale 建议不要动，会影响音质，玄学参数", value=0.4) + vc_submit = gr.Button("转换", variant="primary") + vc_output1 = gr.Textbox(label="Output Message") + vc_output2 = gr.Audio(label="Output Audio") + vc_submit.click(vc_fn, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale], [vc_output1, vc_output2]) + + app.launch() + + + diff --git a/cluster/__init__.py b/cluster/__init__.py new file mode 100644 index 0000000000000000000000000000000000000000..f1b9bde04e73e9218a5d534227caa4c25332f424 --- /dev/null +++ b/cluster/__init__.py @@ -0,0 +1,29 @@ +import numpy as np +import torch +from sklearn.cluster import KMeans + +def get_cluster_model(ckpt_path): + checkpoint = torch.load(ckpt_path) + kmeans_dict = {} + for spk, ckpt in checkpoint.items(): + km = KMeans(ckpt["n_features_in_"]) + km.__dict__["n_features_in_"] = ckpt["n_features_in_"] + km.__dict__["_n_threads"] = ckpt["_n_threads"] + km.__dict__["cluster_centers_"] = ckpt["cluster_centers_"] + kmeans_dict[spk] = km + return kmeans_dict + +def get_cluster_result(model, x, speaker): + """ + x: np.array [t, 256] + return cluster class result + """ + return model[speaker].predict(x) + +def get_cluster_center_result(model, x,speaker): + """x: np.array [t, 256]""" + predict = model[speaker].predict(x) + return model[speaker].cluster_centers_[predict] + +def get_center(model, x,speaker): + return model[speaker].cluster_centers_[x] diff --git a/cluster/train_cluster.py b/cluster/train_cluster.py new file mode 100644 index 0000000000000000000000000000000000000000..4ac025d400414226e66849407f477ae786c3d5d3 --- /dev/null +++ b/cluster/train_cluster.py @@ -0,0 +1,89 @@ +import os +from glob import glob +from pathlib import Path +import torch +import logging +import argparse +import torch +import numpy as np +from sklearn.cluster import KMeans, MiniBatchKMeans +import tqdm +logging.basicConfig(level=logging.INFO) +logger = logging.getLogger(__name__) +import time +import random + +def train_cluster(in_dir, n_clusters, use_minibatch=True, verbose=False): + + logger.info(f"Loading features from {in_dir}") + features = [] + nums = 0 + for path in tqdm.tqdm(in_dir.glob("*.soft.pt")): + features.append(torch.load(path).squeeze(0).numpy().T) + # print(features[-1].shape) + features = np.concatenate(features, axis=0) + print(nums, features.nbytes/ 1024**2, "MB , shape:",features.shape, features.dtype) + features = features.astype(np.float32) + logger.info(f"Clustering features of shape: {features.shape}") + t = time.time() + if use_minibatch: + kmeans = MiniBatchKMeans(n_clusters=n_clusters,verbose=verbose, batch_size=4096, max_iter=80).fit(features) + else: + kmeans = KMeans(n_clusters=n_clusters,verbose=verbose).fit(features) + print(time.time()-t, "s") + + x = { + "n_features_in_": kmeans.n_features_in_, + "_n_threads": kmeans._n_threads, + "cluster_centers_": kmeans.cluster_centers_, + } + print("end") + + return x + + +if __name__ == "__main__": + + parser = argparse.ArgumentParser() + parser.add_argument('--dataset', type=Path, default="./dataset/44k", + help='path of training data directory') + parser.add_argument('--output', type=Path, default="logs/44k", + help='path of model output directory') + + args = parser.parse_args() + + checkpoint_dir = args.output + dataset = args.dataset + n_clusters = 10000 + + ckpt = {} + for spk in os.listdir(dataset): + if os.path.isdir(dataset/spk): + print(f"train kmeans for {spk}...") + in_dir = dataset/spk + x = train_cluster(in_dir, n_clusters, verbose=False) + ckpt[spk] = x + + checkpoint_path = checkpoint_dir / f"kmeans_{n_clusters}.pt" + checkpoint_path.parent.mkdir(exist_ok=True, parents=True) + torch.save( + ckpt, + checkpoint_path, + ) + + + # import cluster + # for spk in tqdm.tqdm(os.listdir("dataset")): + # if os.path.isdir(f"dataset/{spk}"): + # print(f"start kmeans inference for {spk}...") + # for feature_path in tqdm.tqdm(glob(f"dataset/{spk}/*.discrete.npy", recursive=True)): + # mel_path = feature_path.replace(".discrete.npy",".mel.npy") + # mel_spectrogram = np.load(mel_path) + # feature_len = mel_spectrogram.shape[-1] + # c = np.load(feature_path) + # c = utils.tools.repeat_expand_2d(torch.FloatTensor(c), feature_len).numpy() + # feature = c.T + # feature_class = cluster.get_cluster_result(feature, spk) + # np.save(feature_path.replace(".discrete.npy", ".discrete_class.npy"), feature_class) + + diff --git a/configs/config.json b/configs/config.json new file mode 100644 index 0000000000000000000000000000000000000000..0b675ff4eb59ce766ebe8360ffdeec11de656b1a --- /dev/null +++ b/configs/config.json @@ -0,0 +1,94 @@ +{ + "train": { + "log_interval": 200, + "eval_interval": 800, + "seed": 1234, + "epochs": 10000, + "learning_rate": 0.0001, + "betas": [ + 0.8, + 0.99 + ], + "eps": 1e-09, + "batch_size": 16, + "fp16_run": false, + "bf16_run": false, + "lr_decay": 0.999875, + "segment_size": 10240, + "init_lr_ratio": 1, + "warmup_epochs": 0, + "c_mel": 45, + "c_kl": 1.0, + "use_sr": true, + "max_speclen": 512, + "port": "8001", + "keep_ckpts": 3 + }, + "data": { + "training_files": "filelists/44k/train.txt", + "validation_files": "filelists/44k/val.txt", + "max_wav_value": 32768.0, + "sampling_rate": 44100, + "filter_length": 2048, + "hop_length": 512, + "win_length": 2048, + "n_mel_channels": 80, + "mel_fmin": 0.0, + "mel_fmax": 22050 + }, + "model": { + "inter_channels": 192, + "hidden_channels": 192, + "filter_channels": 768, + "n_heads": 2, + "n_layers": 6, + "kernel_size": 3, + "p_dropout": 0.1, + "resblock": "1", + "resblock_kernel_sizes": [ + 3, + 7, + 11 + ], + "resblock_dilation_sizes": [ + [ + 1, + 3, + 5 + ], + [ + 1, + 3, + 5 + ], + [ + 1, + 3, + 5 + ] + ], + "upsample_rates": [ + 8, + 8, + 2, + 2, + 2 + ], + "upsample_initial_channel": 512, + "upsample_kernel_sizes": [ + 16, + 16, + 4, + 4, + 4 + ], + "n_layers_q": 3, + "use_spectral_norm": false, + "gin_channels": 256, + "ssl_dim": 256, + "n_speakers": 200 + }, + "spk": { + "emu": 0 + } +} \ No newline at end of file diff --git a/configs_template/config_template.json b/configs_template/config_template.json new file mode 100644 index 0000000000000000000000000000000000000000..a6555caef49bcb5159ec615adaff41120c93594d --- /dev/null +++ b/configs_template/config_template.json @@ -0,0 +1,66 @@ +{ + "train": { + "log_interval": 200, + "eval_interval": 800, + "seed": 1234, + "epochs": 10000, + "learning_rate": 0.0001, + "betas": [ + 0.8, + 0.99 + ], + "eps": 1e-09, + "batch_size": 6, + "fp16_run": false, + "lr_decay": 0.999875, + "segment_size": 10240, + "init_lr_ratio": 1, + "warmup_epochs": 0, + "c_mel": 45, + "c_kl": 1.0, + "use_sr": true, + "max_speclen": 512, + "port": "8001", + "keep_ckpts": 3, + "all_in_mem": false + }, + "data": { + "training_files": "filelists/train.txt", + "validation_files": "filelists/val.txt", + "max_wav_value": 32768.0, + "sampling_rate": 44100, + "filter_length": 2048, + "hop_length": 512, + "win_length": 2048, + "n_mel_channels": 80, + "mel_fmin": 0.0, + "mel_fmax": 22050 + }, + "model": { + "inter_channels": 192, + "hidden_channels": 192, + "filter_channels": 768, + "n_heads": 2, + "n_layers": 6, + "kernel_size": 3, + "p_dropout": 0.1, + "resblock": "1", + "resblock_kernel_sizes": [3,7,11], + "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]], + "upsample_rates": [ 8, 8, 2, 2, 2], + "upsample_initial_channel": 512, + "upsample_kernel_sizes": [16,16, 4, 4, 4], + "n_layers_q": 3, + "use_spectral_norm": false, + "gin_channels": 256, + "ssl_dim": 256, + "n_speakers": 200 + }, + "spk": { + "nyaru": 0, + "huiyu": 1, + "nen": 2, + "paimon": 3, + "yunhao": 4 + } +} \ No newline at end of file diff --git a/data_utils.py b/data_utils.py new file mode 100644 index 0000000000000000000000000000000000000000..7c76fd1c3a45b8304d916161718c7763874f3e35 --- /dev/null +++ b/data_utils.py @@ -0,0 +1,155 @@ +import time +import os +import random +import numpy as np +import torch +import torch.utils.data + +import modules.commons as commons +import utils +from modules.mel_processing import spectrogram_torch, spec_to_mel_torch +from utils import load_wav_to_torch, load_filepaths_and_text + +# import h5py + + +"""Multi speaker version""" + + +class TextAudioSpeakerLoader(torch.utils.data.Dataset): + """ + 1) loads audio, speaker_id, text pairs + 2) normalizes text and converts them to sequences of integers + 3) computes spectrograms from audio files. + """ + + def __init__(self, audiopaths, hparams, all_in_mem: bool = False): + self.audiopaths = load_filepaths_and_text(audiopaths) + self.max_wav_value = hparams.data.max_wav_value + self.sampling_rate = hparams.data.sampling_rate + self.filter_length = hparams.data.filter_length + self.hop_length = hparams.data.hop_length + self.win_length = hparams.data.win_length + self.sampling_rate = hparams.data.sampling_rate + self.use_sr = hparams.train.use_sr + self.spec_len = hparams.train.max_speclen + self.spk_map = hparams.spk + + random.seed(1234) + random.shuffle(self.audiopaths) + + self.all_in_mem = all_in_mem + if self.all_in_mem: + self.cache = [self.get_audio(p[0]) for p in self.audiopaths] + + def get_audio(self, filename): + filename = filename.replace("\\", "/") + audio, sampling_rate = load_wav_to_torch(filename) + if sampling_rate != self.sampling_rate: + raise ValueError("{} SR doesn't match target {} SR".format( + sampling_rate, self.sampling_rate)) + audio_norm = audio / self.max_wav_value + audio_norm = audio_norm.unsqueeze(0) + spec_filename = filename.replace(".wav", ".spec.pt") + + # Ideally, all data generated after Mar 25 should have .spec.pt + if os.path.exists(spec_filename): + spec = torch.load(spec_filename) + else: + spec = spectrogram_torch(audio_norm, self.filter_length, + self.sampling_rate, self.hop_length, self.win_length, + center=False) + spec = torch.squeeze(spec, 0) + torch.save(spec, spec_filename) + + spk = filename.split("/")[-2] + spk = torch.LongTensor([self.spk_map[spk]]) + + f0 = np.load(filename + ".f0.npy") + f0, uv = utils.interpolate_f0(f0) + f0 = torch.FloatTensor(f0) + uv = torch.FloatTensor(uv) + + c = torch.load(filename+ ".soft.pt") + c = utils.repeat_expand_2d(c.squeeze(0), f0.shape[0]) + + + lmin = min(c.size(-1), spec.size(-1)) + assert abs(c.size(-1) - spec.size(-1)) < 3, (c.size(-1), spec.size(-1), f0.shape, filename) + assert abs(audio_norm.shape[1]-lmin * self.hop_length) < 3 * self.hop_length + spec, c, f0, uv = spec[:, :lmin], c[:, :lmin], f0[:lmin], uv[:lmin] + audio_norm = audio_norm[:, :lmin * self.hop_length] + + return c, f0, spec, audio_norm, spk, uv + + def random_slice(self, c, f0, spec, audio_norm, spk, uv): + # if spec.shape[1] < 30: + # print("skip too short audio:", filename) + # return None + if spec.shape[1] > 800: + start = random.randint(0, spec.shape[1]-800) + end = start + 790 + spec, c, f0, uv = spec[:, start:end], c[:, start:end], f0[start:end], uv[start:end] + audio_norm = audio_norm[:, start * self.hop_length : end * self.hop_length] + + return c, f0, spec, audio_norm, spk, uv + + def __getitem__(self, index): + if self.all_in_mem: + return self.random_slice(*self.cache[index]) + else: + return self.random_slice(*self.get_audio(self.audiopaths[index][0])) + + def __len__(self): + return len(self.audiopaths) + + +class TextAudioCollate: + + def __call__(self, batch): + batch = [b for b in batch if b is not None] + + input_lengths, ids_sorted_decreasing = torch.sort( + torch.LongTensor([x[0].shape[1] for x in batch]), + dim=0, descending=True) + + max_c_len = max([x[0].size(1) for x in batch]) + max_wav_len = max([x[3].size(1) for x in batch]) + + lengths = torch.LongTensor(len(batch)) + + c_padded = torch.FloatTensor(len(batch), batch[0][0].shape[0], max_c_len) + f0_padded = torch.FloatTensor(len(batch), max_c_len) + spec_padded = torch.FloatTensor(len(batch), batch[0][2].shape[0], max_c_len) + wav_padded = torch.FloatTensor(len(batch), 1, max_wav_len) + spkids = torch.LongTensor(len(batch), 1) + uv_padded = torch.FloatTensor(len(batch), max_c_len) + + c_padded.zero_() + spec_padded.zero_() + f0_padded.zero_() + wav_padded.zero_() + uv_padded.zero_() + + for i in range(len(ids_sorted_decreasing)): + row = batch[ids_sorted_decreasing[i]] + + c = row[0] + c_padded[i, :, :c.size(1)] = c + lengths[i] = c.size(1) + + f0 = row[1] + f0_padded[i, :f0.size(0)] = f0 + + spec = row[2] + spec_padded[i, :, :spec.size(1)] = spec + + wav = row[3] + wav_padded[i, :, :wav.size(1)] = wav + + spkids[i, 0] = row[4] + + uv = row[5] + uv_padded[i, :uv.size(0)] = uv + + return c_padded, f0_padded, spec_padded, wav_padded, spkids, lengths, uv_padded diff --git a/dataset_raw/wav_structure.txt b/dataset_raw/wav_structure.txt new file mode 100644 index 0000000000000000000000000000000000000000..68cee4e98b3512989e01945f600fc276e21637e0 --- /dev/null +++ b/dataset_raw/wav_structure.txt @@ -0,0 +1,20 @@ +数据集准备 + +raw +├───speaker0 +│ ├───xxx1-xxx1.wav +│ ├───... +│ └───Lxx-0xx8.wav +└───speaker1 + ├───xx2-0xxx2.wav + ├───... + └───xxx7-xxx007.wav + +此外还需要编辑config.json + +"n_speakers": 10 + +"spk":{ + "speaker0": 0, + "speaker1": 1, +} diff --git a/filelists/test.txt b/filelists/test.txt new file mode 100644 index 0000000000000000000000000000000000000000..be640cffb48b3bc39126f9d1b83a3c992fe6e30d --- /dev/null +++ b/filelists/test.txt @@ -0,0 +1,4 @@ +./dataset/44k/taffy/000562.wav +./dataset/44k/nyaru/000011.wav +./dataset/44k/nyaru/000008.wav +./dataset/44k/taffy/000563.wav diff --git a/filelists/train.txt b/filelists/train.txt new file mode 100644 index 0000000000000000000000000000000000000000..acdb3ccec870a72f0d4da413e6aea97b36331f03 --- /dev/null +++ b/filelists/train.txt @@ -0,0 +1,15 @@ +./dataset/44k/taffy/000549.wav +./dataset/44k/nyaru/000004.wav +./dataset/44k/nyaru/000006.wav +./dataset/44k/taffy/000551.wav +./dataset/44k/nyaru/000009.wav +./dataset/44k/taffy/000561.wav +./dataset/44k/nyaru/000001.wav +./dataset/44k/taffy/000553.wav +./dataset/44k/nyaru/000002.wav +./dataset/44k/taffy/000560.wav +./dataset/44k/taffy/000557.wav +./dataset/44k/nyaru/000005.wav +./dataset/44k/taffy/000554.wav +./dataset/44k/taffy/000550.wav +./dataset/44k/taffy/000559.wav diff --git a/filelists/val.txt b/filelists/val.txt new file mode 100644 index 0000000000000000000000000000000000000000..262dfc97ec1ec3671138954a5c1490add8875b5b --- /dev/null +++ b/filelists/val.txt @@ -0,0 +1,4 @@ +./dataset/44k/nyaru/000003.wav +./dataset/44k/nyaru/000007.wav +./dataset/44k/taffy/000558.wav +./dataset/44k/taffy/000556.wav diff --git a/flask_api.py b/flask_api.py new file mode 100644 index 0000000000000000000000000000000000000000..b3f1e06847b2711a8e5841a4c95375445470d2ee --- /dev/null +++ b/flask_api.py @@ -0,0 +1,60 @@ +import io +import logging + +import soundfile +import torch +import torchaudio +from flask import Flask, request, send_file +from flask_cors import CORS + +from inference.infer_tool import Svc, RealTimeVC + +app = Flask(__name__) + +CORS(app) + +logging.getLogger('numba').setLevel(logging.WARNING) + + +@app.route("/voiceChangeModel", methods=["POST"]) +def voice_change_model(): + request_form = request.form + wave_file = request.files.get("sample", None) + # 变调信息 + f_pitch_change = float(request_form.get("fPitchChange", 0)) + # DAW所需的采样率 + daw_sample = int(float(request_form.get("sampleRate", 0))) + speaker_id = int(float(request_form.get("sSpeakId", 0))) + # http获得wav文件并转换 + input_wav_path = io.BytesIO(wave_file.read()) + + # 模型推理 + if raw_infer: + # out_audio, out_sr = svc_model.infer(speaker_id, f_pitch_change, input_wav_path) + out_audio, out_sr = svc_model.infer(speaker_id, f_pitch_change, input_wav_path, cluster_infer_ratio=0, + auto_predict_f0=False, noice_scale=0.4, f0_filter=False) + tar_audio = torchaudio.functional.resample(out_audio, svc_model.target_sample, daw_sample) + else: + out_audio = svc.process(svc_model, speaker_id, f_pitch_change, input_wav_path, cluster_infer_ratio=0, + auto_predict_f0=False, noice_scale=0.4, f0_filter=False) + tar_audio = torchaudio.functional.resample(torch.from_numpy(out_audio), svc_model.target_sample, daw_sample) + # 返回音频 + out_wav_path = io.BytesIO() + soundfile.write(out_wav_path, tar_audio.cpu().numpy(), daw_sample, format="wav") + out_wav_path.seek(0) + return send_file(out_wav_path, download_name="temp.wav", as_attachment=True) + + +if __name__ == '__main__': + # 启用则为直接切片合成，False为交叉淡化方式 + # vst插件调整0.3-0.5s切片时间可以降低延迟，直接切片方法会有连接处爆音、交叉淡化会有轻微重叠声音 + # 自行选择能接受的方法，或将vst最大切片时间调整为1s，此处设为Ture，延迟大音质稳定一些 + raw_infer = True + # 每个模型和config是唯一对应的 + model_name = "logs/32k/G_174000-Copy1.pth" + config_name = "configs/config.json" + cluster_model_path = "logs/44k/kmeans_10000.pt" + svc_model = Svc(model_name, config_name, cluster_model_path=cluster_model_path) + svc = RealTimeVC() + # 此处与vst插件对应，不建议更改 + app.run(port=6842, host="0.0.0.0", debug=False, threaded=False) diff --git a/flask_api_full_song.py b/flask_api_full_song.py new file mode 100644 index 0000000000000000000000000000000000000000..9dbf66a17114c7f9679717e2938759ae4a371c34 --- /dev/null +++ b/flask_api_full_song.py @@ -0,0 +1,55 @@ +import io +import numpy as np +import soundfile +from flask import Flask, request, send_file + +from inference import infer_tool +from inference import slicer + +app = Flask(__name__) + + +@app.route("/wav2wav", methods=["POST"]) +def wav2wav(): + request_form = request.form + audio_path = request_form.get("audio_path", None) # wav文件地址 + tran = int(float(request_form.get("tran", 0))) # 音调 + spk = request_form.get("spk", 0) # 说话人(id或者name都可以,具体看你的config) + wav_format = request_form.get("wav_format", 'wav') # 范围文件格式 + infer_tool.format_wav(audio_path) + chunks = slicer.cut(audio_path, db_thresh=-40) + audio_data, audio_sr = slicer.chunks2audio(audio_path, chunks) + + audio = [] + for (slice_tag, data) in audio_data: + print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======') + + length = int(np.ceil(len(data) / audio_sr * svc_model.target_sample)) + if slice_tag: + print('jump empty segment') + _audio = np.zeros(length) + else: + # padd + pad_len = int(audio_sr * 0.5) + data = np.concatenate([np.zeros([pad_len]), data, np.zeros([pad_len])]) + raw_path = io.BytesIO() + soundfile.write(raw_path, data, audio_sr, format="wav") + raw_path.seek(0) + out_audio, out_sr = svc_model.infer(spk, tran, raw_path) + svc_model.clear_empty() + _audio = out_audio.cpu().numpy() + pad_len = int(svc_model.target_sample * 0.5) + _audio = _audio[pad_len:-pad_len] + + audio.extend(list(infer_tool.pad_array(_audio, length))) + out_wav_path = io.BytesIO() + soundfile.write(out_wav_path, audio, svc_model.target_sample, format=wav_format) + out_wav_path.seek(0) + return send_file(out_wav_path, download_name=f"temp.{wav_format}", as_attachment=True) + + +if __name__ == '__main__': + model_name = "logs/44k/G_60000.pth" # 模型地址 + config_name = "configs/config.json" # config地址 + svc_model = infer_tool.Svc(model_name, config_name) + app.run(port=1145, host="0.0.0.0", debug=False, threaded=False) diff --git a/hubert/__init__.py b/hubert/__init__.py new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/hubert/hubert_model.py b/hubert/hubert_model.py new file mode 100644 index 0000000000000000000000000000000000000000..7fb642d89b07ca60792debab18e3454f52d8f357 --- /dev/null +++ b/hubert/hubert_model.py @@ -0,0 +1,222 @@ +import copy +import random +from typing import Optional, Tuple + +import torch +import torch.nn as nn +import torch.nn.functional as t_func +from torch.nn.modules.utils import consume_prefix_in_state_dict_if_present + + +class Hubert(nn.Module): + def __init__(self, num_label_embeddings: int = 100, mask: bool = True): + super().__init__() + self._mask = mask + self.feature_extractor = FeatureExtractor() + self.feature_projection = FeatureProjection() + self.positional_embedding = PositionalConvEmbedding() + self.norm = nn.LayerNorm(768) + self.dropout = nn.Dropout(0.1) + self.encoder = TransformerEncoder( + nn.TransformerEncoderLayer( + 768, 12, 3072, activation="gelu", batch_first=True + ), + 12, + ) + self.proj = nn.Linear(768, 256) + + self.masked_spec_embed = nn.Parameter(torch.FloatTensor(768).uniform_()) + self.label_embedding = nn.Embedding(num_label_embeddings, 256) + + def mask(self, x: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]: + mask = None + if self.training and self._mask: + mask = _compute_mask((x.size(0), x.size(1)), 0.8, 10, x.device, 2) + x[mask] = self.masked_spec_embed.to(x.dtype) + return x, mask + + def encode( + self, x: torch.Tensor, layer: Optional[int] = None + ) -> Tuple[torch.Tensor, torch.Tensor]: + x = self.feature_extractor(x) + x = self.feature_projection(x.transpose(1, 2)) + x, mask = self.mask(x) + x = x + self.positional_embedding(x) + x = self.dropout(self.norm(x)) + x = self.encoder(x, output_layer=layer) + return x, mask + + def logits(self, x: torch.Tensor) -> torch.Tensor: + logits = torch.cosine_similarity( + x.unsqueeze(2), + self.label_embedding.weight.unsqueeze(0).unsqueeze(0), + dim=-1, + ) + return logits / 0.1 + + def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]: + x, mask = self.encode(x) + x = self.proj(x) + logits = self.logits(x) + return logits, mask + + +class HubertSoft(Hubert): + def __init__(self): + super().__init__() + + @torch.inference_mode() + def units(self, wav: torch.Tensor) -> torch.Tensor: + wav = t_func.pad(wav, ((400 - 320) // 2, (400 - 320) // 2)) + x, _ = self.encode(wav) + return self.proj(x) + + +class FeatureExtractor(nn.Module): + def __init__(self): + super().__init__() + self.conv0 = nn.Conv1d(1, 512, 10, 5, bias=False) + self.norm0 = nn.GroupNorm(512, 512) + self.conv1 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv2 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv3 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv4 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv5 = nn.Conv1d(512, 512, 2, 2, bias=False) + self.conv6 = nn.Conv1d(512, 512, 2, 2, bias=False) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + x = t_func.gelu(self.norm0(self.conv0(x))) + x = t_func.gelu(self.conv1(x)) + x = t_func.gelu(self.conv2(x)) + x = t_func.gelu(self.conv3(x)) + x = t_func.gelu(self.conv4(x)) + x = t_func.gelu(self.conv5(x)) + x = t_func.gelu(self.conv6(x)) + return x + + +class FeatureProjection(nn.Module): + def __init__(self): + super().__init__() + self.norm = nn.LayerNorm(512) + self.projection = nn.Linear(512, 768) + self.dropout = nn.Dropout(0.1) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + x = self.norm(x) + x = self.projection(x) + x = self.dropout(x) + return x + + +class PositionalConvEmbedding(nn.Module): + def __init__(self): + super().__init__() + self.conv = nn.Conv1d( + 768, + 768, + kernel_size=128, + padding=128 // 2, + groups=16, + ) + self.conv = nn.utils.weight_norm(self.conv, name="weight", dim=2) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + x = self.conv(x.transpose(1, 2)) + x = t_func.gelu(x[:, :, :-1]) + return x.transpose(1, 2) + + +class TransformerEncoder(nn.Module): + def __init__( + self, encoder_layer: nn.TransformerEncoderLayer, num_layers: int + ) -> None: + super(TransformerEncoder, self).__init__() + self.layers = nn.ModuleList( + [copy.deepcopy(encoder_layer) for _ in range(num_layers)] + ) + self.num_layers = num_layers + + def forward( + self, + src: torch.Tensor, + mask: torch.Tensor = None, + src_key_padding_mask: torch.Tensor = None, + output_layer: Optional[int] = None, + ) -> torch.Tensor: + output = src + for layer in self.layers[:output_layer]: + output = layer( + output, src_mask=mask, src_key_padding_mask=src_key_padding_mask + ) + return output + + +def _compute_mask( + shape: Tuple[int, int], + mask_prob: float, + mask_length: int, + device: torch.device, + min_masks: int = 0, +) -> torch.Tensor: + batch_size, sequence_length = shape + + if mask_length < 1: + raise ValueError("`mask_length` has to be bigger than 0.") + + if mask_length > sequence_length: + raise ValueError( + f"`mask_length` has to be smaller than `sequence_length`, but got `mask_length`: {mask_length} and `sequence_length`: {sequence_length}`" + ) + + # compute number of masked spans in batch + num_masked_spans = int(mask_prob * sequence_length / mask_length + random.random()) + num_masked_spans = max(num_masked_spans, min_masks) + + # make sure num masked indices <= sequence_length + if num_masked_spans * mask_length > sequence_length: + num_masked_spans = sequence_length // mask_length + + # SpecAugment mask to fill + mask = torch.zeros((batch_size, sequence_length), device=device, dtype=torch.bool) + + # uniform distribution to sample from, make sure that offset samples are < sequence_length + uniform_dist = torch.ones( + (batch_size, sequence_length - (mask_length - 1)), device=device + ) + + # get random indices to mask + mask_indices = torch.multinomial(uniform_dist, num_masked_spans) + + # expand masked indices to masked spans + mask_indices = ( + mask_indices.unsqueeze(dim=-1) + .expand((batch_size, num_masked_spans, mask_length)) + .reshape(batch_size, num_masked_spans * mask_length) + ) + offsets = ( + torch.arange(mask_length, device=device)[None, None, :] + .expand((batch_size, num_masked_spans, mask_length)) + .reshape(batch_size, num_masked_spans * mask_length) + ) + mask_idxs = mask_indices + offsets + + # scatter indices to mask + mask = mask.scatter(1, mask_idxs, True) + + return mask + + +def hubert_soft( + path: str, +) -> HubertSoft: + r"""HuBERT-Soft from `"A Comparison of Discrete and Soft Speech Units for Improved Voice Conversion"`. + Args: + path (str): path of a pretrained model + """ + hubert = HubertSoft() + checkpoint = torch.load(path) + consume_prefix_in_state_dict_if_present(checkpoint, "module.") + hubert.load_state_dict(checkpoint) + hubert.eval() + return hubert diff --git a/hubert/hubert_model_onnx.py b/hubert/hubert_model_onnx.py new file mode 100644 index 0000000000000000000000000000000000000000..d18f3c2a0fc29592a573a9780308d38f059640b9 --- /dev/null +++ b/hubert/hubert_model_onnx.py @@ -0,0 +1,217 @@ +import copy +import random +from typing import Optional, Tuple + +import torch +import torch.nn as nn +import torch.nn.functional as t_func +from torch.nn.modules.utils import consume_prefix_in_state_dict_if_present + + +class Hubert(nn.Module): + def __init__(self, num_label_embeddings: int = 100, mask: bool = True): + super().__init__() + self._mask = mask + self.feature_extractor = FeatureExtractor() + self.feature_projection = FeatureProjection() + self.positional_embedding = PositionalConvEmbedding() + self.norm = nn.LayerNorm(768) + self.dropout = nn.Dropout(0.1) + self.encoder = TransformerEncoder( + nn.TransformerEncoderLayer( + 768, 12, 3072, activation="gelu", batch_first=True + ), + 12, + ) + self.proj = nn.Linear(768, 256) + + self.masked_spec_embed = nn.Parameter(torch.FloatTensor(768).uniform_()) + self.label_embedding = nn.Embedding(num_label_embeddings, 256) + + def mask(self, x: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]: + mask = None + if self.training and self._mask: + mask = _compute_mask((x.size(0), x.size(1)), 0.8, 10, x.device, 2) + x[mask] = self.masked_spec_embed.to(x.dtype) + return x, mask + + def encode( + self, x: torch.Tensor, layer: Optional[int] = None + ) -> Tuple[torch.Tensor, torch.Tensor]: + x = self.feature_extractor(x) + x = self.feature_projection(x.transpose(1, 2)) + x, mask = self.mask(x) + x = x + self.positional_embedding(x) + x = self.dropout(self.norm(x)) + x = self.encoder(x, output_layer=layer) + return x, mask + + def logits(self, x: torch.Tensor) -> torch.Tensor: + logits = torch.cosine_similarity( + x.unsqueeze(2), + self.label_embedding.weight.unsqueeze(0).unsqueeze(0), + dim=-1, + ) + return logits / 0.1 + + +class HubertSoft(Hubert): + def __init__(self): + super().__init__() + + def units(self, wav: torch.Tensor) -> torch.Tensor: + wav = t_func.pad(wav, ((400 - 320) // 2, (400 - 320) // 2)) + x, _ = self.encode(wav) + return self.proj(x) + + def forward(self, x): + return self.units(x) + +class FeatureExtractor(nn.Module): + def __init__(self): + super().__init__() + self.conv0 = nn.Conv1d(1, 512, 10, 5, bias=False) + self.norm0 = nn.GroupNorm(512, 512) + self.conv1 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv2 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv3 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv4 = nn.Conv1d(512, 512, 3, 2, bias=False) + self.conv5 = nn.Conv1d(512, 512, 2, 2, bias=False) + self.conv6 = nn.Conv1d(512, 512, 2, 2, bias=False) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + x = t_func.gelu(self.norm0(self.conv0(x))) + x = t_func.gelu(self.conv1(x)) + x = t_func.gelu(self.conv2(x)) + x = t_func.gelu(self.conv3(x)) + x = t_func.gelu(self.conv4(x)) + x = t_func.gelu(self.conv5(x)) + x = t_func.gelu(self.conv6(x)) + return x + + +class FeatureProjection(nn.Module): + def __init__(self): + super().__init__() + self.norm = nn.LayerNorm(512) + self.projection = nn.Linear(512, 768) + self.dropout = nn.Dropout(0.1) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + x = self.norm(x) + x = self.projection(x) + x = self.dropout(x) + return x + + +class PositionalConvEmbedding(nn.Module): + def __init__(self): + super().__init__() + self.conv = nn.Conv1d( + 768, + 768, + kernel_size=128, + padding=128 // 2, + groups=16, + ) + self.conv = nn.utils.weight_norm(self.conv, name="weight", dim=2) + + def forward(self, x: torch.Tensor) -> torch.Tensor: + x = self.conv(x.transpose(1, 2)) + x = t_func.gelu(x[:, :, :-1]) + return x.transpose(1, 2) + + +class TransformerEncoder(nn.Module): + def __init__( + self, encoder_layer: nn.TransformerEncoderLayer, num_layers: int + ) -> None: + super(TransformerEncoder, self).__init__() + self.layers = nn.ModuleList( + [copy.deepcopy(encoder_layer) for _ in range(num_layers)] + ) + self.num_layers = num_layers + + def forward( + self, + src: torch.Tensor, + mask: torch.Tensor = None, + src_key_padding_mask: torch.Tensor = None, + output_layer: Optional[int] = None, + ) -> torch.Tensor: + output = src + for layer in self.layers[:output_layer]: + output = layer( + output, src_mask=mask, src_key_padding_mask=src_key_padding_mask + ) + return output + + +def _compute_mask( + shape: Tuple[int, int], + mask_prob: float, + mask_length: int, + device: torch.device, + min_masks: int = 0, +) -> torch.Tensor: + batch_size, sequence_length = shape + + if mask_length < 1: + raise ValueError("`mask_length` has to be bigger than 0.") + + if mask_length > sequence_length: + raise ValueError( + f"`mask_length` has to be smaller than `sequence_length`, but got `mask_length`: {mask_length} and `sequence_length`: {sequence_length}`" + ) + + # compute number of masked spans in batch + num_masked_spans = int(mask_prob * sequence_length / mask_length + random.random()) + num_masked_spans = max(num_masked_spans, min_masks) + + # make sure num masked indices <= sequence_length + if num_masked_spans * mask_length > sequence_length: + num_masked_spans = sequence_length // mask_length + + # SpecAugment mask to fill + mask = torch.zeros((batch_size, sequence_length), device=device, dtype=torch.bool) + + # uniform distribution to sample from, make sure that offset samples are < sequence_length + uniform_dist = torch.ones( + (batch_size, sequence_length - (mask_length - 1)), device=device + ) + + # get random indices to mask + mask_indices = torch.multinomial(uniform_dist, num_masked_spans) + + # expand masked indices to masked spans + mask_indices = ( + mask_indices.unsqueeze(dim=-1) + .expand((batch_size, num_masked_spans, mask_length)) + .reshape(batch_size, num_masked_spans * mask_length) + ) + offsets = ( + torch.arange(mask_length, device=device)[None, None, :] + .expand((batch_size, num_masked_spans, mask_length)) + .reshape(batch_size, num_masked_spans * mask_length) + ) + mask_idxs = mask_indices + offsets + + # scatter indices to mask + mask = mask.scatter(1, mask_idxs, True) + + return mask + + +def hubert_soft( + path: str, +) -> HubertSoft: + r"""HuBERT-Soft from `"A Comparison of Discrete and Soft Speech Units for Improved Voice Conversion"`. + Args: + path (str): path of a pretrained model + """ + hubert = HubertSoft() + checkpoint = torch.load(path) + consume_prefix_in_state_dict_if_present(checkpoint, "module.") + hubert.load_state_dict(checkpoint) + hubert.eval() + return hubert diff --git a/hubert/put_hubert_ckpt_here b/hubert/put_hubert_ckpt_here new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/inference/__init__.py b/inference/__init__.py new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/inference/infer_tool.py b/inference/infer_tool.py new file mode 100644 index 0000000000000000000000000000000000000000..5328c549bfcfa789a74e56729219e5607a6612a6 --- /dev/null +++ b/inference/infer_tool.py @@ -0,0 +1,340 @@ +import hashlib +import io +import json +import logging +import os +import time +from pathlib import Path +from inference import slicer + +import librosa +import numpy as np +# import onnxruntime +import parselmouth +import soundfile +import torch +import torchaudio + +import cluster +from hubert import hubert_model +import utils +from models import SynthesizerTrn + +logging.getLogger('matplotlib').setLevel(logging.WARNING) + + +def read_temp(file_name): + if not os.path.exists(file_name): + with open(file_name, "w") as f: + f.write(json.dumps({"info": "temp_dict"})) + return {} + else: + try: + with open(file_name, "r") as f: + data = f.read() + data_dict = json.loads(data) + if os.path.getsize(file_name) > 50 * 1024 * 1024: + f_name = file_name.replace("\\", "/").split("/")[-1] + print(f"clean {f_name}") + for wav_hash in list(data_dict.keys()): + if int(time.time()) - int(data_dict[wav_hash]["time"]) > 14 * 24 * 3600: + del data_dict[wav_hash] + except Exception as e: + print(e) + print(f"{file_name} error,auto rebuild file") + data_dict = {"info": "temp_dict"} + return data_dict + + +def write_temp(file_name, data): + with open(file_name, "w") as f: + f.write(json.dumps(data)) + + +def timeit(func): + def run(*args, **kwargs): + t = time.time() + res = func(*args, **kwargs) + print('executing \'%s\' costed %.3fs' % (func.__name__, time.time() - t)) + return res + + return run + + +def format_wav(audio_path): + if Path(audio_path).suffix == '.wav': + return + raw_audio, raw_sample_rate = librosa.load(audio_path, mono=True, sr=None) + soundfile.write(Path(audio_path).with_suffix(".wav"), raw_audio, raw_sample_rate) + + +def get_end_file(dir_path, end): + file_lists = [] + for root, dirs, files in os.walk(dir_path): + files = [f for f in files if f[0] != '.'] + dirs[:] = [d for d in dirs if d[0] != '.'] + for f_file in files: + if f_file.endswith(end): + file_lists.append(os.path.join(root, f_file).replace("\\", "/")) + return file_lists + + +def get_md5(content): + return hashlib.new("md5", content).hexdigest() + +def fill_a_to_b(a, b): + if len(a) < len(b): + for _ in range(0, len(b) - len(a)): + a.append(a[0]) + +def mkdir(paths: list): + for path in paths: + if not os.path.exists(path): + os.mkdir(path) + +def pad_array(arr, target_length): + current_length = arr.shape[0] + if current_length >= target_length: + return arr + else: + pad_width = target_length - current_length + pad_left = pad_width // 2 + pad_right = pad_width - pad_left + padded_arr = np.pad(arr, (pad_left, pad_right), 'constant', constant_values=(0, 0)) + return padded_arr + +def split_list_by_n(list_collection, n, pre=0): + for i in range(0, len(list_collection), n): + yield list_collection[i-pre if i-pre>=0 else i: i + n] + + +class F0FilterException(Exception): + pass + +class Svc(object): + def __init__(self, net_g_path, config_path, + device=None, + cluster_model_path="logs/44k/kmeans_10000.pt", + nsf_hifigan_enhance = False + ): + self.net_g_path = net_g_path + if device is None: + self.dev = torch.device("cuda" if torch.cuda.is_available() else "cpu") + else: + self.dev = torch.device(device) + self.net_g_ms = None + self.hps_ms = utils.get_hparams_from_file(config_path) + self.target_sample = self.hps_ms.data.sampling_rate + self.hop_size = self.hps_ms.data.hop_length + self.spk2id = self.hps_ms.spk + self.nsf_hifigan_enhance = nsf_hifigan_enhance + # 加载hubert + self.hubert_model = utils.get_hubert_model().to(self.dev) + self.load_model() + if os.path.exists(cluster_model_path): + self.cluster_model = cluster.get_cluster_model(cluster_model_path) + if self.nsf_hifigan_enhance: + from modules.enhancer import Enhancer + self.enhancer = Enhancer('nsf-hifigan', 'pretrain/nsf_hifigan/model',device=self.dev) + + def load_model(self): + # 获取模型配置 + self.net_g_ms = SynthesizerTrn( + self.hps_ms.data.filter_length // 2 + 1, + self.hps_ms.train.segment_size // self.hps_ms.data.hop_length, + **self.hps_ms.model) + _ = utils.load_checkpoint(self.net_g_path, self.net_g_ms, None) + if "half" in self.net_g_path and torch.cuda.is_available(): + _ = self.net_g_ms.half().eval().to(self.dev) + else: + _ = self.net_g_ms.eval().to(self.dev) + + + + def get_unit_f0(self, in_path, tran, cluster_infer_ratio, speaker, f0_filter ,F0_mean_pooling): + + wav, sr = librosa.load(in_path, sr=self.target_sample) + + if F0_mean_pooling == True: + f0, uv = utils.compute_f0_uv_torchcrepe(torch.FloatTensor(wav), sampling_rate=self.target_sample, hop_length=self.hop_size,device=self.dev) + if f0_filter and sum(f0) == 0: + raise F0FilterException("未检测到人声") + f0 = torch.FloatTensor(list(f0)) + uv = torch.FloatTensor(list(uv)) + if F0_mean_pooling == False: + f0 = utils.compute_f0_parselmouth(wav, sampling_rate=self.target_sample, hop_length=self.hop_size) + if f0_filter and sum(f0) == 0: + raise F0FilterException("未检测到人声") + f0, uv = utils.interpolate_f0(f0) + f0 = torch.FloatTensor(f0) + uv = torch.FloatTensor(uv) + + f0 = f0 * 2 ** (tran / 12) + f0 = f0.unsqueeze(0).to(self.dev) + uv = uv.unsqueeze(0).to(self.dev) + + wav16k = librosa.resample(wav, orig_sr=self.target_sample, target_sr=16000) + wav16k = torch.from_numpy(wav16k).to(self.dev) + c = utils.get_hubert_content(self.hubert_model, wav_16k_tensor=wav16k) + c = utils.repeat_expand_2d(c.squeeze(0), f0.shape[1]) + + if cluster_infer_ratio !=0: + cluster_c = cluster.get_cluster_center_result(self.cluster_model, c.cpu().numpy().T, speaker).T + cluster_c = torch.FloatTensor(cluster_c).to(self.dev) + c = cluster_infer_ratio * cluster_c + (1 - cluster_infer_ratio) * c + + c = c.unsqueeze(0) + return c, f0, uv + + def infer(self, speaker, tran, raw_path, + cluster_infer_ratio=0, + auto_predict_f0=False, + noice_scale=0.4, + f0_filter=False, + F0_mean_pooling=False, + enhancer_adaptive_key = 0 + ): + + speaker_id = self.spk2id.__dict__.get(speaker) + if not speaker_id and type(speaker) is int: + if len(self.spk2id.__dict__) >= speaker: + speaker_id = speaker + sid = torch.LongTensor([int(speaker_id)]).to(self.dev).unsqueeze(0) + c, f0, uv = self.get_unit_f0(raw_path, tran, cluster_infer_ratio, speaker, f0_filter,F0_mean_pooling) + if "half" in self.net_g_path and torch.cuda.is_available(): + c = c.half() + with torch.no_grad(): + start = time.time() + audio = self.net_g_ms.infer(c, f0=f0, g=sid, uv=uv, predict_f0=auto_predict_f0, noice_scale=noice_scale)[0,0].data.float() + if self.nsf_hifigan_enhance: + audio, _ = self.enhancer.enhance( + audio[None,:], + self.target_sample, + f0[:,:,None], + self.hps_ms.data.hop_length, + adaptive_key = enhancer_adaptive_key) + use_time = time.time() - start + print("vits use time:{}".format(use_time)) + return audio, audio.shape[-1] + + def clear_empty(self): + # 清理显存 + torch.cuda.empty_cache() + + def slice_inference(self, + raw_audio_path, + spk, + tran, + slice_db, + cluster_infer_ratio, + auto_predict_f0, + noice_scale, + pad_seconds=0.5, + clip_seconds=0, + lg_num=0, + lgr_num =0.75, + F0_mean_pooling = False, + enhancer_adaptive_key = 0 + ): + wav_path = raw_audio_path + chunks = slicer.cut(wav_path, db_thresh=slice_db) + audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks) + per_size = int(clip_seconds*audio_sr) + lg_size = int(lg_num*audio_sr) + lg_size_r = int(lg_size*lgr_num) + lg_size_c_l = (lg_size-lg_size_r)//2 + lg_size_c_r = lg_size-lg_size_r-lg_size_c_l + lg = np.linspace(0,1,lg_size_r) if lg_size!=0 else 0 + + audio = [] + for (slice_tag, data) in audio_data: + print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======') + # padd + length = int(np.ceil(len(data) / audio_sr * self.target_sample)) + if slice_tag: + print('jump empty segment') + _audio = np.zeros(length) + audio.extend(list(pad_array(_audio, length))) + continue + if per_size != 0: + datas = split_list_by_n(data, per_size,lg_size) + else: + datas = [data] + for k,dat in enumerate(datas): + per_length = int(np.ceil(len(dat) / audio_sr * self.target_sample)) if clip_seconds!=0 else length + if clip_seconds!=0: print(f'###=====segment clip start, {round(len(dat) / audio_sr, 3)}s======') + # padd + pad_len = int(audio_sr * pad_seconds) + dat = np.concatenate([np.zeros([pad_len]), dat, np.zeros([pad_len])]) + raw_path = io.BytesIO() + soundfile.write(raw_path, dat, audio_sr, format="wav") + raw_path.seek(0) + out_audio, out_sr = self.infer(spk, tran, raw_path, + cluster_infer_ratio=cluster_infer_ratio, + auto_predict_f0=auto_predict_f0, + noice_scale=noice_scale, + F0_mean_pooling = F0_mean_pooling, + enhancer_adaptive_key = enhancer_adaptive_key + ) + _audio = out_audio.cpu().numpy() + pad_len = int(self.target_sample * pad_seconds) + _audio = _audio[pad_len:-pad_len] + _audio = pad_array(_audio, per_length) + if lg_size!=0 and k!=0: + lg1 = audio[-(lg_size_r+lg_size_c_r):-lg_size_c_r] if lgr_num != 1 else audio[-lg_size:] + lg2 = _audio[lg_size_c_l:lg_size_c_l+lg_size_r] if lgr_num != 1 else _audio[0:lg_size] + lg_pre = lg1*(1-lg)+lg2*lg + audio = audio[0:-(lg_size_r+lg_size_c_r)] if lgr_num != 1 else audio[0:-lg_size] + audio.extend(lg_pre) + _audio = _audio[lg_size_c_l+lg_size_r:] if lgr_num != 1 else _audio[lg_size:] + audio.extend(list(_audio)) + return np.array(audio) + +class RealTimeVC: + def __init__(self): + self.last_chunk = None + self.last_o = None + self.chunk_len = 16000 # 区块长度 + self.pre_len = 3840 # 交叉淡化长度，640的倍数 + + """输入输出都是1维numpy 音频波形数组""" + + def process(self, svc_model, speaker_id, f_pitch_change, input_wav_path, + cluster_infer_ratio=0, + auto_predict_f0=False, + noice_scale=0.4, + f0_filter=False): + + import maad + audio, sr = torchaudio.load(input_wav_path) + audio = audio.cpu().numpy()[0] + temp_wav = io.BytesIO() + if self.last_chunk is None: + input_wav_path.seek(0) + + audio, sr = svc_model.infer(speaker_id, f_pitch_change, input_wav_path, + cluster_infer_ratio=cluster_infer_ratio, + auto_predict_f0=auto_predict_f0, + noice_scale=noice_scale, + f0_filter=f0_filter) + + audio = audio.cpu().numpy() + self.last_chunk = audio[-self.pre_len:] + self.last_o = audio + return audio[-self.chunk_len:] + else: + audio = np.concatenate([self.last_chunk, audio]) + soundfile.write(temp_wav, audio, sr, format="wav") + temp_wav.seek(0) + + audio, sr = svc_model.infer(speaker_id, f_pitch_change, temp_wav, + cluster_infer_ratio=cluster_infer_ratio, + auto_predict_f0=auto_predict_f0, + noice_scale=noice_scale, + f0_filter=f0_filter) + + audio = audio.cpu().numpy() + ret = maad.util.crossfade(self.last_o, audio, self.pre_len) + self.last_chunk = audio[-self.pre_len:] + self.last_o = audio + return ret[self.chunk_len:2 * self.chunk_len] diff --git a/inference/infer_tool_grad.py b/inference/infer_tool_grad.py new file mode 100644 index 0000000000000000000000000000000000000000..b75af49c08e2e724839828bc419792ed580809bb --- /dev/null +++ b/inference/infer_tool_grad.py @@ -0,0 +1,160 @@ +import hashlib +import json +import logging +import os +import time +from pathlib import Path +import io +import librosa +import maad +import numpy as np +from inference import slicer +import parselmouth +import soundfile +import torch +import torchaudio + +from hubert import hubert_model +import utils +from models import SynthesizerTrn +logging.getLogger('numba').setLevel(logging.WARNING) +logging.getLogger('matplotlib').setLevel(logging.WARNING) + +def resize2d_f0(x, target_len): + source = np.array(x) + source[source < 0.001] = np.nan + target = np.interp(np.arange(0, len(source) * target_len, len(source)) / target_len, np.arange(0, len(source)), + source) + res = np.nan_to_num(target) + return res + +def get_f0(x, p_len,f0_up_key=0): + + time_step = 160 / 16000 * 1000 + f0_min = 50 + f0_max = 1100 + f0_mel_min = 1127 * np.log(1 + f0_min / 700) + f0_mel_max = 1127 * np.log(1 + f0_max / 700) + + f0 = parselmouth.Sound(x, 16000).to_pitch_ac( + time_step=time_step / 1000, voicing_threshold=0.6, + pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency'] + + pad_size=(p_len - len(f0) + 1) // 2 + if(pad_size>0 or p_len - len(f0) - pad_size>0): + f0 = np.pad(f0,[[pad_size,p_len - len(f0) - pad_size]], mode='constant') + + f0 *= pow(2, f0_up_key / 12) + f0_mel = 1127 * np.log(1 + f0 / 700) + f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (f0_mel_max - f0_mel_min) + 1 + f0_mel[f0_mel <= 1] = 1 + f0_mel[f0_mel > 255] = 255 + f0_coarse = np.rint(f0_mel).astype(np.int) + return f0_coarse, f0 + +def clean_pitch(input_pitch): + num_nan = np.sum(input_pitch == 1) + if num_nan / len(input_pitch) > 0.9: + input_pitch[input_pitch != 1] = 1 + return input_pitch + + +def plt_pitch(input_pitch): + input_pitch = input_pitch.astype(float) + input_pitch[input_pitch == 1] = np.nan + return input_pitch + + +def f0_to_pitch(ff): + f0_pitch = 69 + 12 * np.log2(ff / 440) + return f0_pitch + + +def fill_a_to_b(a, b): + if len(a) < len(b): + for _ in range(0, len(b) - len(a)): + a.append(a[0]) + + +def mkdir(paths: list): + for path in paths: + if not os.path.exists(path): + os.mkdir(path) + + +class VitsSvc(object): + def __init__(self): + self.device = torch.device("cuda" if torch.cuda.is_available() else "cpu") + self.SVCVITS = None + self.hps = None + self.speakers = None + self.hubert_soft = utils.get_hubert_model() + + def set_device(self, device): + self.device = torch.device(device) + self.hubert_soft.to(self.device) + if self.SVCVITS != None: + self.SVCVITS.to(self.device) + + def loadCheckpoint(self, path): + self.hps = utils.get_hparams_from_file(f"checkpoints/{path}/config.json") + self.SVCVITS = SynthesizerTrn( + self.hps.data.filter_length // 2 + 1, + self.hps.train.segment_size // self.hps.data.hop_length, + **self.hps.model) + _ = utils.load_checkpoint(f"checkpoints/{path}/model.pth", self.SVCVITS, None) + _ = self.SVCVITS.eval().to(self.device) + self.speakers = self.hps.spk + + def get_units(self, source, sr): + source = source.unsqueeze(0).to(self.device) + with torch.inference_mode(): + units = self.hubert_soft.units(source) + return units + + + def get_unit_pitch(self, in_path, tran): + source, sr = torchaudio.load(in_path) + source = torchaudio.functional.resample(source, sr, 16000) + if len(source.shape) == 2 and source.shape[1] >= 2: + source = torch.mean(source, dim=0).unsqueeze(0) + soft = self.get_units(source, sr).squeeze(0).cpu().numpy() + f0_coarse, f0 = get_f0(source.cpu().numpy()[0], soft.shape[0]*2, tran) + return soft, f0 + + def infer(self, speaker_id, tran, raw_path): + speaker_id = self.speakers[speaker_id] + sid = torch.LongTensor([int(speaker_id)]).to(self.device).unsqueeze(0) + soft, pitch = self.get_unit_pitch(raw_path, tran) + f0 = torch.FloatTensor(clean_pitch(pitch)).unsqueeze(0).to(self.device) + stn_tst = torch.FloatTensor(soft) + with torch.no_grad(): + x_tst = stn_tst.unsqueeze(0).to(self.device) + x_tst = torch.repeat_interleave(x_tst, repeats=2, dim=1).transpose(1, 2) + audio = self.SVCVITS.infer(x_tst, f0=f0, g=sid)[0,0].data.float() + return audio, audio.shape[-1] + + def inference(self,srcaudio,chara,tran,slice_db): + sampling_rate, audio = srcaudio + audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32) + if len(audio.shape) > 1: + audio = librosa.to_mono(audio.transpose(1, 0)) + if sampling_rate != 16000: + audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000) + soundfile.write("tmpwav.wav", audio, 16000, format="wav") + chunks = slicer.cut("tmpwav.wav", db_thresh=slice_db) + audio_data, audio_sr = slicer.chunks2audio("tmpwav.wav", chunks) + audio = [] + for (slice_tag, data) in audio_data: + length = int(np.ceil(len(data) / audio_sr * self.hps.data.sampling_rate)) + raw_path = io.BytesIO() + soundfile.write(raw_path, data, audio_sr, format="wav") + raw_path.seek(0) + if slice_tag: + _audio = np.zeros(length) + else: + out_audio, out_sr = self.infer(chara, tran, raw_path) + _audio = out_audio.cpu().numpy() + audio.extend(list(_audio)) + audio = (np.array(audio) * 32768.0).astype('int16') + return (self.hps.data.sampling_rate,audio) diff --git a/inference/slicer.py b/inference/slicer.py new file mode 100644 index 0000000000000000000000000000000000000000..b05840bcf6bdced0b6e2adbecb1a1dd5b3dee462 --- /dev/null +++ b/inference/slicer.py @@ -0,0 +1,142 @@ +import librosa +import torch +import torchaudio + + +class Slicer: + def __init__(self, + sr: int, + threshold: float = -40., + min_length: int = 5000, + min_interval: int = 300, + hop_size: int = 20, + max_sil_kept: int = 5000): + if not min_length >= min_interval >= hop_size: + raise ValueError('The following condition must be satisfied: min_length >= min_interval >= hop_size') + if not max_sil_kept >= hop_size: + raise ValueError('The following condition must be satisfied: max_sil_kept >= hop_size') + min_interval = sr * min_interval / 1000 + self.threshold = 10 ** (threshold / 20.) + self.hop_size = round(sr * hop_size / 1000) + self.win_size = min(round(min_interval), 4 * self.hop_size) + self.min_length = round(sr * min_length / 1000 / self.hop_size) + self.min_interval = round(min_interval / self.hop_size) + self.max_sil_kept = round(sr * max_sil_kept / 1000 / self.hop_size) + + def _apply_slice(self, waveform, begin, end): + if len(waveform.shape) > 1: + return waveform[:, begin * self.hop_size: min(waveform.shape[1], end * self.hop_size)] + else: + return waveform[begin * self.hop_size: min(waveform.shape[0], end * self.hop_size)] + + # @timeit + def slice(self, waveform): + if len(waveform.shape) > 1: + samples = librosa.to_mono(waveform) + else: + samples = waveform + if samples.shape[0] <= self.min_length: + return {"0": {"slice": False, "split_time": f"0,{len(waveform)}"}} + rms_list = librosa.feature.rms(y=samples, frame_length=self.win_size, hop_length=self.hop_size).squeeze(0) + sil_tags = [] + silence_start = None + clip_start = 0 + for i, rms in enumerate(rms_list): + # Keep looping while frame is silent. + if rms < self.threshold: + # Record start of silent frames. + if silence_start is None: + silence_start = i + continue + # Keep looping while frame is not silent and silence start has not been recorded. + if silence_start is None: + continue + # Clear recorded silence start if interval is not enough or clip is too short + is_leading_silence = silence_start == 0 and i > self.max_sil_kept + need_slice_middle = i - silence_start >= self.min_interval and i - clip_start >= self.min_length + if not is_leading_silence and not need_slice_middle: + silence_start = None + continue + # Need slicing. Record the range of silent frames to be removed. + if i - silence_start <= self.max_sil_kept: + pos = rms_list[silence_start: i + 1].argmin() + silence_start + if silence_start == 0: + sil_tags.append((0, pos)) + else: + sil_tags.append((pos, pos)) + clip_start = pos + elif i - silence_start <= self.max_sil_kept * 2: + pos = rms_list[i - self.max_sil_kept: silence_start + self.max_sil_kept + 1].argmin() + pos += i - self.max_sil_kept + pos_l = rms_list[silence_start: silence_start + self.max_sil_kept + 1].argmin() + silence_start + pos_r = rms_list[i - self.max_sil_kept: i + 1].argmin() + i - self.max_sil_kept + if silence_start == 0: + sil_tags.append((0, pos_r)) + clip_start = pos_r + else: + sil_tags.append((min(pos_l, pos), max(pos_r, pos))) + clip_start = max(pos_r, pos) + else: + pos_l = rms_list[silence_start: silence_start + self.max_sil_kept + 1].argmin() + silence_start + pos_r = rms_list[i - self.max_sil_kept: i + 1].argmin() + i - self.max_sil_kept + if silence_start == 0: + sil_tags.append((0, pos_r)) + else: + sil_tags.append((pos_l, pos_r)) + clip_start = pos_r + silence_start = None + # Deal with trailing silence. + total_frames = rms_list.shape[0] + if silence_start is not None and total_frames - silence_start >= self.min_interval: + silence_end = min(total_frames, silence_start + self.max_sil_kept) + pos = rms_list[silence_start: silence_end + 1].argmin() + silence_start + sil_tags.append((pos, total_frames + 1)) + # Apply and return slices. + if len(sil_tags) == 0: + return {"0": {"slice": False, "split_time": f"0,{len(waveform)}"}} + else: + chunks = [] + # 第一段静音并非从头开始，补上有声片段 + if sil_tags[0][0]: + chunks.append( + {"slice": False, "split_time": f"0,{min(waveform.shape[0], sil_tags[0][0] * self.hop_size)}"}) + for i in range(0, len(sil_tags)): + # 标识有声片段（跳过第一段） + if i: + chunks.append({"slice": False, + "split_time": f"{sil_tags[i - 1][1] * self.hop_size},{min(waveform.shape[0], sil_tags[i][0] * self.hop_size)}"}) + # 标识所有静音片段 + chunks.append({"slice": True, + "split_time": f"{sil_tags[i][0] * self.hop_size},{min(waveform.shape[0], sil_tags[i][1] * self.hop_size)}"}) + # 最后一段静音并非结尾，补上结尾片段 + if sil_tags[-1][1] * self.hop_size < len(waveform): + chunks.append({"slice": False, "split_time": f"{sil_tags[-1][1] * self.hop_size},{len(waveform)}"}) + chunk_dict = {} + for i in range(len(chunks)): + chunk_dict[str(i)] = chunks[i] + return chunk_dict + + +def cut(audio_path, db_thresh=-30, min_len=5000): + audio, sr = librosa.load(audio_path, sr=None) + slicer = Slicer( + sr=sr, + threshold=db_thresh, + min_length=min_len + ) + chunks = slicer.slice(audio) + return chunks + + +def chunks2audio(audio_path, chunks): + chunks = dict(chunks) + audio, sr = torchaudio.load(audio_path) + if len(audio.shape) == 2 and audio.shape[1] >= 2: + audio = torch.mean(audio, dim=0).unsqueeze(0) + audio = audio.cpu().numpy()[0] + result = [] + for k, v in chunks.items(): + tag = v["split_time"].split(",") + if tag[0] != tag[1]: + result.append((v["slice"], audio[int(tag[0]):int(tag[1])])) + return result, sr diff --git a/inference_main.py b/inference_main.py new file mode 100644 index 0000000000000000000000000000000000000000..b6c9ff8fc771c1bada0b04d59f0af4c87a524089 --- /dev/null +++ b/inference_main.py @@ -0,0 +1,137 @@ +import io +import logging +import time +from pathlib import Path + +import librosa +import matplotlib.pyplot as plt +import numpy as np +import soundfile + +from inference import infer_tool +from inference import slicer +from inference.infer_tool import Svc + +logging.getLogger('numba').setLevel(logging.WARNING) +chunks_dict = infer_tool.read_temp("inference/chunks_temp.json") + + + +def main(): + import argparse + + parser = argparse.ArgumentParser(description='sovits4 inference') + + # 一定要设置的部分 + parser.add_argument('-m', '--model_path', type=str, default="logs/44k/G_0.pth", help='模型路径') + parser.add_argument('-c', '--config_path', type=str, default="configs/config.json", help='配置文件路径') + parser.add_argument('-cl', '--clip', type=float, default=0, help='音频强制切片，默认0为自动切片，单位为秒/s') + parser.add_argument('-n', '--clean_names', type=str, nargs='+', default=["君の知らない物語-src.wav"], help='wav文件名列表，放在raw文件夹下') + parser.add_argument('-t', '--trans', type=int, nargs='+', default=[0], help='音高调整，支持正负（半音）') + parser.add_argument('-s', '--spk_list', type=str, nargs='+', default=['nen'], help='合成目标说话人名称') + + # 可选项部分 + parser.add_argument('-a', '--auto_predict_f0', action='store_true', default=False,help='语音转换自动预测音高，转换歌声时不要打开这个会严重跑调') + parser.add_argument('-cm', '--cluster_model_path', type=str, default="logs/44k/kmeans_10000.pt", help='聚类模型路径，如果没有训练聚类则随便填') + parser.add_argument('-cr', '--cluster_infer_ratio', type=float, default=0, help='聚类方案占比，范围0-1，若没有训练聚类模型则默认0即可') + parser.add_argument('-lg', '--linear_gradient', type=float, default=0, help='两段音频切片的交叉淡入长度，如果强制切片后出现人声不连贯可调整该数值，如果连贯建议采用默认值0，单位为秒') + parser.add_argument('-fmp', '--f0_mean_pooling', type=bool, default=False, help='是否对F0使用均值滤波器(池化)，对部分哑音有改善。注意，启动该选项会导致推理速度下降，默认关闭') + parser.add_argument('-eh', '--enhance', type=bool, default=False, help='是否使用NSF_HIFIGAN增强器,该选项对部分训练集少的模型有一定的音质增强效果，但是对训练好的模型有反面效果，默认关闭') + + # 不用动的部分 + parser.add_argument('-sd', '--slice_db', type=int, default=-40, help='默认-40，嘈杂的音频可以-30，干声保留呼吸可以-50') + parser.add_argument('-d', '--device', type=str, default=None, help='推理设备，None则为自动选择cpu和gpu') + parser.add_argument('-ns', '--noice_scale', type=float, default=0.4, help='噪音级别，会影响咬字和音质，较为玄学') + parser.add_argument('-p', '--pad_seconds', type=float, default=0.5, help='推理音频pad秒数，由于未知原因开头结尾会有异响，pad一小段静音段后就不会出现') + parser.add_argument('-wf', '--wav_format', type=str, default='flac', help='音频输出格式') + parser.add_argument('-lgr', '--linear_gradient_retain', type=float, default=0.75, help='自动音频切片后，需要舍弃每段切片的头尾。该参数设置交叉长度保留的比例，范围0-1,左开右闭') + parser.add_argument('-eak', '--enhancer_adaptive_key', type=int, default=0, help='使增强器适应更高的音域(单位为半音数)|默认为0') + + args = parser.parse_args() + + clean_names = args.clean_names + trans = args.trans + spk_list = args.spk_list + slice_db = args.slice_db + wav_format = args.wav_format + auto_predict_f0 = args.auto_predict_f0 + cluster_infer_ratio = args.cluster_infer_ratio + noice_scale = args.noice_scale + pad_seconds = args.pad_seconds + clip = args.clip + lg = args.linear_gradient + lgr = args.linear_gradient_retain + F0_mean_pooling = args.f0_mean_pooling + enhance = args.enhance + enhancer_adaptive_key = args.enhancer_adaptive_key + + svc_model = Svc(args.model_path, args.config_path, args.device, args.cluster_model_path,enhance) + infer_tool.mkdir(["raw", "results"]) + + infer_tool.fill_a_to_b(trans, clean_names) + for clean_name, tran in zip(clean_names, trans): + raw_audio_path = f"raw/{clean_name}" + if "." not in raw_audio_path: + raw_audio_path += ".wav" + infer_tool.format_wav(raw_audio_path) + wav_path = Path(raw_audio_path).with_suffix('.wav') + chunks = slicer.cut(wav_path, db_thresh=slice_db) + audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks) + per_size = int(clip*audio_sr) + lg_size = int(lg*audio_sr) + lg_size_r = int(lg_size*lgr) + lg_size_c_l = (lg_size-lg_size_r)//2 + lg_size_c_r = lg_size-lg_size_r-lg_size_c_l + lg = np.linspace(0,1,lg_size_r) if lg_size!=0 else 0 + + for spk in spk_list: + audio = [] + for (slice_tag, data) in audio_data: + print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======') + + length = int(np.ceil(len(data) / audio_sr * svc_model.target_sample)) + if slice_tag: + print('jump empty segment') + _audio = np.zeros(length) + audio.extend(list(infer_tool.pad_array(_audio, length))) + continue + if per_size != 0: + datas = infer_tool.split_list_by_n(data, per_size,lg_size) + else: + datas = [data] + for k,dat in enumerate(datas): + per_length = int(np.ceil(len(dat) / audio_sr * svc_model.target_sample)) if clip!=0 else length + if clip!=0: print(f'###=====segment clip start, {round(len(dat) / audio_sr, 3)}s======') + # padd + pad_len = int(audio_sr * pad_seconds) + dat = np.concatenate([np.zeros([pad_len]), dat, np.zeros([pad_len])]) + raw_path = io.BytesIO() + soundfile.write(raw_path, dat, audio_sr, format="wav") + raw_path.seek(0) + out_audio, out_sr = svc_model.infer(spk, tran, raw_path, + cluster_infer_ratio=cluster_infer_ratio, + auto_predict_f0=auto_predict_f0, + noice_scale=noice_scale, + F0_mean_pooling = F0_mean_pooling, + enhancer_adaptive_key = enhancer_adaptive_key + ) + _audio = out_audio.cpu().numpy() + pad_len = int(svc_model.target_sample * pad_seconds) + _audio = _audio[pad_len:-pad_len] + _audio = infer_tool.pad_array(_audio, per_length) + if lg_size!=0 and k!=0: + lg1 = audio[-(lg_size_r+lg_size_c_r):-lg_size_c_r] if lgr != 1 else audio[-lg_size:] + lg2 = _audio[lg_size_c_l:lg_size_c_l+lg_size_r] if lgr != 1 else _audio[0:lg_size] + lg_pre = lg1*(1-lg)+lg2*lg + audio = audio[0:-(lg_size_r+lg_size_c_r)] if lgr != 1 else audio[0:-lg_size] + audio.extend(lg_pre) + _audio = _audio[lg_size_c_l+lg_size_r:] if lgr != 1 else _audio[lg_size:] + audio.extend(list(_audio)) + key = "auto" if auto_predict_f0 else f"{tran}key" + cluster_name = "" if cluster_infer_ratio == 0 else f"_{cluster_infer_ratio}" + res_path = f'./results/{clean_name}_{key}_{spk}{cluster_name}.{wav_format}' + soundfile.write(res_path, audio, svc_model.target_sample, format=wav_format) + svc_model.clear_empty() + +if __name__ == '__main__': + main() diff --git a/logs/44k/put_pretrained_model_here b/logs/44k/put_pretrained_model_here new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/models.py b/models.py new file mode 100644 index 0000000000000000000000000000000000000000..13278d680493970f5a670cf3fc955a6e9b7ab1d5 --- /dev/null +++ b/models.py @@ -0,0 +1,420 @@ +import copy +import math +import torch +from torch import nn +from torch.nn import functional as F + +import modules.attentions as attentions +import modules.commons as commons +import modules.modules as modules + +from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d +from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm + +import utils +from modules.commons import init_weights, get_padding +from vdecoder.hifigan.models import Generator +from utils import f0_to_coarse + +class ResidualCouplingBlock(nn.Module): + def __init__(self, + channels, + hidden_channels, + kernel_size, + dilation_rate, + n_layers, + n_flows=4, + gin_channels=0): + super().__init__() + self.channels = channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.dilation_rate = dilation_rate + self.n_layers = n_layers + self.n_flows = n_flows + self.gin_channels = gin_channels + + self.flows = nn.ModuleList() + for i in range(n_flows): + self.flows.append(modules.ResidualCouplingLayer(channels, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels, mean_only=True)) + self.flows.append(modules.Flip()) + + def forward(self, x, x_mask, g=None, reverse=False): + if not reverse: + for flow in self.flows: + x, _ = flow(x, x_mask, g=g, reverse=reverse) + else: + for flow in reversed(self.flows): + x = flow(x, x_mask, g=g, reverse=reverse) + return x + + +class Encoder(nn.Module): + def __init__(self, + in_channels, + out_channels, + hidden_channels, + kernel_size, + dilation_rate, + n_layers, + gin_channels=0): + super().__init__() + self.in_channels = in_channels + self.out_channels = out_channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.dilation_rate = dilation_rate + self.n_layers = n_layers + self.gin_channels = gin_channels + + self.pre = nn.Conv1d(in_channels, hidden_channels, 1) + self.enc = modules.WN(hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels) + self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1) + + def forward(self, x, x_lengths, g=None): + # print(x.shape,x_lengths.shape) + x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype) + x = self.pre(x) * x_mask + x = self.enc(x, x_mask, g=g) + stats = self.proj(x) * x_mask + m, logs = torch.split(stats, self.out_channels, dim=1) + z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask + return z, m, logs, x_mask + + +class TextEncoder(nn.Module): + def __init__(self, + out_channels, + hidden_channels, + kernel_size, + n_layers, + gin_channels=0, + filter_channels=None, + n_heads=None, + p_dropout=None): + super().__init__() + self.out_channels = out_channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.n_layers = n_layers + self.gin_channels = gin_channels + self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1) + self.f0_emb = nn.Embedding(256, hidden_channels) + + self.enc_ = attentions.Encoder( + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout) + + def forward(self, x, x_mask, f0=None, noice_scale=1): + x = x + self.f0_emb(f0).transpose(1,2) + x = self.enc_(x * x_mask, x_mask) + stats = self.proj(x) * x_mask + m, logs = torch.split(stats, self.out_channels, dim=1) + z = (m + torch.randn_like(m) * torch.exp(logs) * noice_scale) * x_mask + + return z, m, logs, x_mask + + + +class DiscriminatorP(torch.nn.Module): + def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False): + super(DiscriminatorP, self).__init__() + self.period = period + self.use_spectral_norm = use_spectral_norm + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(get_padding(kernel_size, 1), 0))), + ]) + self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0))) + + def forward(self, x): + fmap = [] + + # 1d to 2d + b, c, t = x.shape + if t % self.period != 0: # pad first + n_pad = self.period - (t % self.period) + x = F.pad(x, (0, n_pad), "reflect") + t = t + n_pad + x = x.view(b, c, t // self.period, self.period) + + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, modules.LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class DiscriminatorS(torch.nn.Module): + def __init__(self, use_spectral_norm=False): + super(DiscriminatorS, self).__init__() + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv1d(1, 16, 15, 1, padding=7)), + norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)), + norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)), + norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)), + norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)), + norm_f(Conv1d(1024, 1024, 5, 1, padding=2)), + ]) + self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1)) + + def forward(self, x): + fmap = [] + + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, modules.LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class MultiPeriodDiscriminator(torch.nn.Module): + def __init__(self, use_spectral_norm=False): + super(MultiPeriodDiscriminator, self).__init__() + periods = [2,3,5,7,11] + + discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)] + discs = discs + [DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods] + self.discriminators = nn.ModuleList(discs) + + def forward(self, y, y_hat): + y_d_rs = [] + y_d_gs = [] + fmap_rs = [] + fmap_gs = [] + for i, d in enumerate(self.discriminators): + y_d_r, fmap_r = d(y) + y_d_g, fmap_g = d(y_hat) + y_d_rs.append(y_d_r) + y_d_gs.append(y_d_g) + fmap_rs.append(fmap_r) + fmap_gs.append(fmap_g) + + return y_d_rs, y_d_gs, fmap_rs, fmap_gs + + +class SpeakerEncoder(torch.nn.Module): + def __init__(self, mel_n_channels=80, model_num_layers=3, model_hidden_size=256, model_embedding_size=256): + super(SpeakerEncoder, self).__init__() + self.lstm = nn.LSTM(mel_n_channels, model_hidden_size, model_num_layers, batch_first=True) + self.linear = nn.Linear(model_hidden_size, model_embedding_size) + self.relu = nn.ReLU() + + def forward(self, mels): + self.lstm.flatten_parameters() + _, (hidden, _) = self.lstm(mels) + embeds_raw = self.relu(self.linear(hidden[-1])) + return embeds_raw / torch.norm(embeds_raw, dim=1, keepdim=True) + + def compute_partial_slices(self, total_frames, partial_frames, partial_hop): + mel_slices = [] + for i in range(0, total_frames-partial_frames, partial_hop): + mel_range = torch.arange(i, i+partial_frames) + mel_slices.append(mel_range) + + return mel_slices + + def embed_utterance(self, mel, partial_frames=128, partial_hop=64): + mel_len = mel.size(1) + last_mel = mel[:,-partial_frames:] + + if mel_len > partial_frames: + mel_slices = self.compute_partial_slices(mel_len, partial_frames, partial_hop) + mels = list(mel[:,s] for s in mel_slices) + mels.append(last_mel) + mels = torch.stack(tuple(mels), 0).squeeze(1) + + with torch.no_grad(): + partial_embeds = self(mels) + embed = torch.mean(partial_embeds, axis=0).unsqueeze(0) + #embed = embed / torch.linalg.norm(embed, 2) + else: + with torch.no_grad(): + embed = self(last_mel) + + return embed + +class F0Decoder(nn.Module): + def __init__(self, + out_channels, + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout, + spk_channels=0): + super().__init__() + self.out_channels = out_channels + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.spk_channels = spk_channels + + self.prenet = nn.Conv1d(hidden_channels, hidden_channels, 3, padding=1) + self.decoder = attentions.FFT( + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout) + self.proj = nn.Conv1d(hidden_channels, out_channels, 1) + self.f0_prenet = nn.Conv1d(1, hidden_channels , 3, padding=1) + self.cond = nn.Conv1d(spk_channels, hidden_channels, 1) + + def forward(self, x, norm_f0, x_mask, spk_emb=None): + x = torch.detach(x) + if (spk_emb is not None): + x = x + self.cond(spk_emb) + x += self.f0_prenet(norm_f0) + x = self.prenet(x) * x_mask + x = self.decoder(x * x_mask, x_mask) + x = self.proj(x) * x_mask + return x + + +class SynthesizerTrn(nn.Module): + """ + Synthesizer for Training + """ + + def __init__(self, + spec_channels, + segment_size, + inter_channels, + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout, + resblock, + resblock_kernel_sizes, + resblock_dilation_sizes, + upsample_rates, + upsample_initial_channel, + upsample_kernel_sizes, + gin_channels, + ssl_dim, + n_speakers, + sampling_rate=44100, + **kwargs): + + super().__init__() + self.spec_channels = spec_channels + self.inter_channels = inter_channels + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.resblock = resblock + self.resblock_kernel_sizes = resblock_kernel_sizes + self.resblock_dilation_sizes = resblock_dilation_sizes + self.upsample_rates = upsample_rates + self.upsample_initial_channel = upsample_initial_channel + self.upsample_kernel_sizes = upsample_kernel_sizes + self.segment_size = segment_size + self.gin_channels = gin_channels + self.ssl_dim = ssl_dim + self.emb_g = nn.Embedding(n_speakers, gin_channels) + + self.pre = nn.Conv1d(ssl_dim, hidden_channels, kernel_size=5, padding=2) + + self.enc_p = TextEncoder( + inter_channels, + hidden_channels, + filter_channels=filter_channels, + n_heads=n_heads, + n_layers=n_layers, + kernel_size=kernel_size, + p_dropout=p_dropout + ) + hps = { + "sampling_rate": sampling_rate, + "inter_channels": inter_channels, + "resblock": resblock, + "resblock_kernel_sizes": resblock_kernel_sizes, + "resblock_dilation_sizes": resblock_dilation_sizes, + "upsample_rates": upsample_rates, + "upsample_initial_channel": upsample_initial_channel, + "upsample_kernel_sizes": upsample_kernel_sizes, + "gin_channels": gin_channels, + } + self.dec = Generator(h=hps) + self.enc_q = Encoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16, gin_channels=gin_channels) + self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels) + self.f0_decoder = F0Decoder( + 1, + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout, + spk_channels=gin_channels + ) + self.emb_uv = nn.Embedding(2, hidden_channels) + + def forward(self, c, f0, uv, spec, g=None, c_lengths=None, spec_lengths=None): + g = self.emb_g(g).transpose(1,2) + # ssl prenet + x_mask = torch.unsqueeze(commons.sequence_mask(c_lengths, c.size(2)), 1).to(c.dtype) + x = self.pre(c) * x_mask + self.emb_uv(uv.long()).transpose(1,2) + + # f0 predict + lf0 = 2595. * torch.log10(1. + f0.unsqueeze(1) / 700.) / 500 + norm_lf0 = utils.normalize_f0(lf0, x_mask, uv) + pred_lf0 = self.f0_decoder(x, norm_lf0, x_mask, spk_emb=g) + + # encoder + z_ptemp, m_p, logs_p, _ = self.enc_p(x, x_mask, f0=f0_to_coarse(f0)) + z, m_q, logs_q, spec_mask = self.enc_q(spec, spec_lengths, g=g) + + # flow + z_p = self.flow(z, spec_mask, g=g) + z_slice, pitch_slice, ids_slice = commons.rand_slice_segments_with_pitch(z, f0, spec_lengths, self.segment_size) + + # nsf decoder + o = self.dec(z_slice, g=g, f0=pitch_slice) + + return o, ids_slice, spec_mask, (z, z_p, m_p, logs_p, m_q, logs_q), pred_lf0, norm_lf0, lf0 + + def infer(self, c, f0, uv, g=None, noice_scale=0.35, predict_f0=False): + c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device) + g = self.emb_g(g).transpose(1,2) + x_mask = torch.unsqueeze(commons.sequence_mask(c_lengths, c.size(2)), 1).to(c.dtype) + x = self.pre(c) * x_mask + self.emb_uv(uv.long()).transpose(1,2) + + if predict_f0: + lf0 = 2595. * torch.log10(1. + f0.unsqueeze(1) / 700.) / 500 + norm_lf0 = utils.normalize_f0(lf0, x_mask, uv, random_scale=False) + pred_lf0 = self.f0_decoder(x, norm_lf0, x_mask, spk_emb=g) + f0 = (700 * (torch.pow(10, pred_lf0 * 500 / 2595) - 1)).squeeze(1) + + z_p, m_p, logs_p, c_mask = self.enc_p(x, x_mask, f0=f0_to_coarse(f0), noice_scale=noice_scale) + z = self.flow(z_p, c_mask, g=g, reverse=True) + o = self.dec(z * c_mask, g=g, f0=f0) + return o diff --git a/modules/__init__.py b/modules/__init__.py new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/modules/attentions.py b/modules/attentions.py new file mode 100644 index 0000000000000000000000000000000000000000..f9c11ca4a3acb86bf1abc04d9dcfa82a4ed4061f --- /dev/null +++ b/modules/attentions.py @@ -0,0 +1,349 @@ +import copy +import math +import numpy as np +import torch +from torch import nn +from torch.nn import functional as F + +import modules.commons as commons +import modules.modules as modules +from modules.modules import LayerNorm + + +class FFT(nn.Module): + def __init__(self, hidden_channels, filter_channels, n_heads, n_layers=1, kernel_size=1, p_dropout=0., + proximal_bias=False, proximal_init=True, **kwargs): + super().__init__() + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.proximal_bias = proximal_bias + self.proximal_init = proximal_init + + self.drop = nn.Dropout(p_dropout) + self.self_attn_layers = nn.ModuleList() + self.norm_layers_0 = nn.ModuleList() + self.ffn_layers = nn.ModuleList() + self.norm_layers_1 = nn.ModuleList() + for i in range(self.n_layers): + self.self_attn_layers.append( + MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout, proximal_bias=proximal_bias, + proximal_init=proximal_init)) + self.norm_layers_0.append(LayerNorm(hidden_channels)) + self.ffn_layers.append( + FFN(hidden_channels, hidden_channels, filter_channels, kernel_size, p_dropout=p_dropout, causal=True)) + self.norm_layers_1.append(LayerNorm(hidden_channels)) + + def forward(self, x, x_mask): + """ + x: decoder input + h: encoder output + """ + self_attn_mask = commons.subsequent_mask(x_mask.size(2)).to(device=x.device, dtype=x.dtype) + x = x * x_mask + for i in range(self.n_layers): + y = self.self_attn_layers[i](x, x, self_attn_mask) + y = self.drop(y) + x = self.norm_layers_0[i](x + y) + + y = self.ffn_layers[i](x, x_mask) + y = self.drop(y) + x = self.norm_layers_1[i](x + y) + x = x * x_mask + return x + + +class Encoder(nn.Module): + def __init__(self, hidden_channels, filter_channels, n_heads, n_layers, kernel_size=1, p_dropout=0., window_size=4, **kwargs): + super().__init__() + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.window_size = window_size + + self.drop = nn.Dropout(p_dropout) + self.attn_layers = nn.ModuleList() + self.norm_layers_1 = nn.ModuleList() + self.ffn_layers = nn.ModuleList() + self.norm_layers_2 = nn.ModuleList() + for i in range(self.n_layers): + self.attn_layers.append(MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout, window_size=window_size)) + self.norm_layers_1.append(LayerNorm(hidden_channels)) + self.ffn_layers.append(FFN(hidden_channels, hidden_channels, filter_channels, kernel_size, p_dropout=p_dropout)) + self.norm_layers_2.append(LayerNorm(hidden_channels)) + + def forward(self, x, x_mask): + attn_mask = x_mask.unsqueeze(2) * x_mask.unsqueeze(-1) + x = x * x_mask + for i in range(self.n_layers): + y = self.attn_layers[i](x, x, attn_mask) + y = self.drop(y) + x = self.norm_layers_1[i](x + y) + + y = self.ffn_layers[i](x, x_mask) + y = self.drop(y) + x = self.norm_layers_2[i](x + y) + x = x * x_mask + return x + + +class Decoder(nn.Module): + def __init__(self, hidden_channels, filter_channels, n_heads, n_layers, kernel_size=1, p_dropout=0., proximal_bias=False, proximal_init=True, **kwargs): + super().__init__() + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.proximal_bias = proximal_bias + self.proximal_init = proximal_init + + self.drop = nn.Dropout(p_dropout) + self.self_attn_layers = nn.ModuleList() + self.norm_layers_0 = nn.ModuleList() + self.encdec_attn_layers = nn.ModuleList() + self.norm_layers_1 = nn.ModuleList() + self.ffn_layers = nn.ModuleList() + self.norm_layers_2 = nn.ModuleList() + for i in range(self.n_layers): + self.self_attn_layers.append(MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout, proximal_bias=proximal_bias, proximal_init=proximal_init)) + self.norm_layers_0.append(LayerNorm(hidden_channels)) + self.encdec_attn_layers.append(MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout)) + self.norm_layers_1.append(LayerNorm(hidden_channels)) + self.ffn_layers.append(FFN(hidden_channels, hidden_channels, filter_channels, kernel_size, p_dropout=p_dropout, causal=True)) + self.norm_layers_2.append(LayerNorm(hidden_channels)) + + def forward(self, x, x_mask, h, h_mask): + """ + x: decoder input + h: encoder output + """ + self_attn_mask = commons.subsequent_mask(x_mask.size(2)).to(device=x.device, dtype=x.dtype) + encdec_attn_mask = h_mask.unsqueeze(2) * x_mask.unsqueeze(-1) + x = x * x_mask + for i in range(self.n_layers): + y = self.self_attn_layers[i](x, x, self_attn_mask) + y = self.drop(y) + x = self.norm_layers_0[i](x + y) + + y = self.encdec_attn_layers[i](x, h, encdec_attn_mask) + y = self.drop(y) + x = self.norm_layers_1[i](x + y) + + y = self.ffn_layers[i](x, x_mask) + y = self.drop(y) + x = self.norm_layers_2[i](x + y) + x = x * x_mask + return x + + +class MultiHeadAttention(nn.Module): + def __init__(self, channels, out_channels, n_heads, p_dropout=0., window_size=None, heads_share=True, block_length=None, proximal_bias=False, proximal_init=False): + super().__init__() + assert channels % n_heads == 0 + + self.channels = channels + self.out_channels = out_channels + self.n_heads = n_heads + self.p_dropout = p_dropout + self.window_size = window_size + self.heads_share = heads_share + self.block_length = block_length + self.proximal_bias = proximal_bias + self.proximal_init = proximal_init + self.attn = None + + self.k_channels = channels // n_heads + self.conv_q = nn.Conv1d(channels, channels, 1) + self.conv_k = nn.Conv1d(channels, channels, 1) + self.conv_v = nn.Conv1d(channels, channels, 1) + self.conv_o = nn.Conv1d(channels, out_channels, 1) + self.drop = nn.Dropout(p_dropout) + + if window_size is not None: + n_heads_rel = 1 if heads_share else n_heads + rel_stddev = self.k_channels**-0.5 + self.emb_rel_k = nn.Parameter(torch.randn(n_heads_rel, window_size * 2 + 1, self.k_channels) * rel_stddev) + self.emb_rel_v = nn.Parameter(torch.randn(n_heads_rel, window_size * 2 + 1, self.k_channels) * rel_stddev) + + nn.init.xavier_uniform_(self.conv_q.weight) + nn.init.xavier_uniform_(self.conv_k.weight) + nn.init.xavier_uniform_(self.conv_v.weight) + if proximal_init: + with torch.no_grad(): + self.conv_k.weight.copy_(self.conv_q.weight) + self.conv_k.bias.copy_(self.conv_q.bias) + + def forward(self, x, c, attn_mask=None): + q = self.conv_q(x) + k = self.conv_k(c) + v = self.conv_v(c) + + x, self.attn = self.attention(q, k, v, mask=attn_mask) + + x = self.conv_o(x) + return x + + def attention(self, query, key, value, mask=None): + # reshape [b, d, t] -> [b, n_h, t, d_k] + b, d, t_s, t_t = (*key.size(), query.size(2)) + query = query.view(b, self.n_heads, self.k_channels, t_t).transpose(2, 3) + key = key.view(b, self.n_heads, self.k_channels, t_s).transpose(2, 3) + value = value.view(b, self.n_heads, self.k_channels, t_s).transpose(2, 3) + + scores = torch.matmul(query / math.sqrt(self.k_channels), key.transpose(-2, -1)) + if self.window_size is not None: + assert t_s == t_t, "Relative attention is only available for self-attention." + key_relative_embeddings = self._get_relative_embeddings(self.emb_rel_k, t_s) + rel_logits = self._matmul_with_relative_keys(query /math.sqrt(self.k_channels), key_relative_embeddings) + scores_local = self._relative_position_to_absolute_position(rel_logits) + scores = scores + scores_local + if self.proximal_bias: + assert t_s == t_t, "Proximal bias is only available for self-attention." + scores = scores + self._attention_bias_proximal(t_s).to(device=scores.device, dtype=scores.dtype) + if mask is not None: + scores = scores.masked_fill(mask == 0, -1e4) + if self.block_length is not None: + assert t_s == t_t, "Local attention is only available for self-attention." + block_mask = torch.ones_like(scores).triu(-self.block_length).tril(self.block_length) + scores = scores.masked_fill(block_mask == 0, -1e4) + p_attn = F.softmax(scores, dim=-1) # [b, n_h, t_t, t_s] + p_attn = self.drop(p_attn) + output = torch.matmul(p_attn, value) + if self.window_size is not None: + relative_weights = self._absolute_position_to_relative_position(p_attn) + value_relative_embeddings = self._get_relative_embeddings(self.emb_rel_v, t_s) + output = output + self._matmul_with_relative_values(relative_weights, value_relative_embeddings) + output = output.transpose(2, 3).contiguous().view(b, d, t_t) # [b, n_h, t_t, d_k] -> [b, d, t_t] + return output, p_attn + + def _matmul_with_relative_values(self, x, y): + """ + x: [b, h, l, m] + y: [h or 1, m, d] + ret: [b, h, l, d] + """ + ret = torch.matmul(x, y.unsqueeze(0)) + return ret + + def _matmul_with_relative_keys(self, x, y): + """ + x: [b, h, l, d] + y: [h or 1, m, d] + ret: [b, h, l, m] + """ + ret = torch.matmul(x, y.unsqueeze(0).transpose(-2, -1)) + return ret + + def _get_relative_embeddings(self, relative_embeddings, length): + max_relative_position = 2 * self.window_size + 1 + # Pad first before slice to avoid using cond ops. + pad_length = max(length - (self.window_size + 1), 0) + slice_start_position = max((self.window_size + 1) - length, 0) + slice_end_position = slice_start_position + 2 * length - 1 + if pad_length > 0: + padded_relative_embeddings = F.pad( + relative_embeddings, + commons.convert_pad_shape([[0, 0], [pad_length, pad_length], [0, 0]])) + else: + padded_relative_embeddings = relative_embeddings + used_relative_embeddings = padded_relative_embeddings[:,slice_start_position:slice_end_position] + return used_relative_embeddings + + def _relative_position_to_absolute_position(self, x): + """ + x: [b, h, l, 2*l-1] + ret: [b, h, l, l] + """ + batch, heads, length, _ = x.size() + # Concat columns of pad to shift from relative to absolute indexing. + x = F.pad(x, commons.convert_pad_shape([[0,0],[0,0],[0,0],[0,1]])) + + # Concat extra elements so to add up to shape (len+1, 2*len-1). + x_flat = x.view([batch, heads, length * 2 * length]) + x_flat = F.pad(x_flat, commons.convert_pad_shape([[0,0],[0,0],[0,length-1]])) + + # Reshape and slice out the padded elements. + x_final = x_flat.view([batch, heads, length+1, 2*length-1])[:, :, :length, length-1:] + return x_final + + def _absolute_position_to_relative_position(self, x): + """ + x: [b, h, l, l] + ret: [b, h, l, 2*l-1] + """ + batch, heads, length, _ = x.size() + # padd along column + x = F.pad(x, commons.convert_pad_shape([[0, 0], [0, 0], [0, 0], [0, length-1]])) + x_flat = x.view([batch, heads, length**2 + length*(length -1)]) + # add 0's in the beginning that will skew the elements after reshape + x_flat = F.pad(x_flat, commons.convert_pad_shape([[0, 0], [0, 0], [length, 0]])) + x_final = x_flat.view([batch, heads, length, 2*length])[:,:,:,1:] + return x_final + + def _attention_bias_proximal(self, length): + """Bias for self-attention to encourage attention to close positions. + Args: + length: an integer scalar. + Returns: + a Tensor with shape [1, 1, length, length] + """ + r = torch.arange(length, dtype=torch.float32) + diff = torch.unsqueeze(r, 0) - torch.unsqueeze(r, 1) + return torch.unsqueeze(torch.unsqueeze(-torch.log1p(torch.abs(diff)), 0), 0) + + +class FFN(nn.Module): + def __init__(self, in_channels, out_channels, filter_channels, kernel_size, p_dropout=0., activation=None, causal=False): + super().__init__() + self.in_channels = in_channels + self.out_channels = out_channels + self.filter_channels = filter_channels + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.activation = activation + self.causal = causal + + if causal: + self.padding = self._causal_padding + else: + self.padding = self._same_padding + + self.conv_1 = nn.Conv1d(in_channels, filter_channels, kernel_size) + self.conv_2 = nn.Conv1d(filter_channels, out_channels, kernel_size) + self.drop = nn.Dropout(p_dropout) + + def forward(self, x, x_mask): + x = self.conv_1(self.padding(x * x_mask)) + if self.activation == "gelu": + x = x * torch.sigmoid(1.702 * x) + else: + x = torch.relu(x) + x = self.drop(x) + x = self.conv_2(self.padding(x * x_mask)) + return x * x_mask + + def _causal_padding(self, x): + if self.kernel_size == 1: + return x + pad_l = self.kernel_size - 1 + pad_r = 0 + padding = [[0, 0], [0, 0], [pad_l, pad_r]] + x = F.pad(x, commons.convert_pad_shape(padding)) + return x + + def _same_padding(self, x): + if self.kernel_size == 1: + return x + pad_l = (self.kernel_size - 1) // 2 + pad_r = self.kernel_size // 2 + padding = [[0, 0], [0, 0], [pad_l, pad_r]] + x = F.pad(x, commons.convert_pad_shape(padding)) + return x diff --git a/modules/commons.py b/modules/commons.py new file mode 100644 index 0000000000000000000000000000000000000000..074888006392e956ce204d8368362dbb2cd4e304 --- /dev/null +++ b/modules/commons.py @@ -0,0 +1,188 @@ +import math +import numpy as np +import torch +from torch import nn +from torch.nn import functional as F + +def slice_pitch_segments(x, ids_str, segment_size=4): + ret = torch.zeros_like(x[:, :segment_size]) + for i in range(x.size(0)): + idx_str = ids_str[i] + idx_end = idx_str + segment_size + ret[i] = x[i, idx_str:idx_end] + return ret + +def rand_slice_segments_with_pitch(x, pitch, x_lengths=None, segment_size=4): + b, d, t = x.size() + if x_lengths is None: + x_lengths = t + ids_str_max = x_lengths - segment_size + 1 + ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long) + ret = slice_segments(x, ids_str, segment_size) + ret_pitch = slice_pitch_segments(pitch, ids_str, segment_size) + return ret, ret_pitch, ids_str + +def init_weights(m, mean=0.0, std=0.01): + classname = m.__class__.__name__ + if classname.find("Conv") != -1: + m.weight.data.normal_(mean, std) + + +def get_padding(kernel_size, dilation=1): + return int((kernel_size*dilation - dilation)/2) + + +def convert_pad_shape(pad_shape): + l = pad_shape[::-1] + pad_shape = [item for sublist in l for item in sublist] + return pad_shape + + +def intersperse(lst, item): + result = [item] * (len(lst) * 2 + 1) + result[1::2] = lst + return result + + +def kl_divergence(m_p, logs_p, m_q, logs_q): + """KL(P||Q)""" + kl = (logs_q - logs_p) - 0.5 + kl += 0.5 * (torch.exp(2. * logs_p) + ((m_p - m_q)**2)) * torch.exp(-2. * logs_q) + return kl + + +def rand_gumbel(shape): + """Sample from the Gumbel distribution, protect from overflows.""" + uniform_samples = torch.rand(shape) * 0.99998 + 0.00001 + return -torch.log(-torch.log(uniform_samples)) + + +def rand_gumbel_like(x): + g = rand_gumbel(x.size()).to(dtype=x.dtype, device=x.device) + return g + + +def slice_segments(x, ids_str, segment_size=4): + ret = torch.zeros_like(x[:, :, :segment_size]) + for i in range(x.size(0)): + idx_str = ids_str[i] + idx_end = idx_str + segment_size + ret[i] = x[i, :, idx_str:idx_end] + return ret + + +def rand_slice_segments(x, x_lengths=None, segment_size=4): + b, d, t = x.size() + if x_lengths is None: + x_lengths = t + ids_str_max = x_lengths - segment_size + 1 + ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long) + ret = slice_segments(x, ids_str, segment_size) + return ret, ids_str + + +def rand_spec_segments(x, x_lengths=None, segment_size=4): + b, d, t = x.size() + if x_lengths is None: + x_lengths = t + ids_str_max = x_lengths - segment_size + ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long) + ret = slice_segments(x, ids_str, segment_size) + return ret, ids_str + + +def get_timing_signal_1d( + length, channels, min_timescale=1.0, max_timescale=1.0e4): + position = torch.arange(length, dtype=torch.float) + num_timescales = channels // 2 + log_timescale_increment = ( + math.log(float(max_timescale) / float(min_timescale)) / + (num_timescales - 1)) + inv_timescales = min_timescale * torch.exp( + torch.arange(num_timescales, dtype=torch.float) * -log_timescale_increment) + scaled_time = position.unsqueeze(0) * inv_timescales.unsqueeze(1) + signal = torch.cat([torch.sin(scaled_time), torch.cos(scaled_time)], 0) + signal = F.pad(signal, [0, 0, 0, channels % 2]) + signal = signal.view(1, channels, length) + return signal + + +def add_timing_signal_1d(x, min_timescale=1.0, max_timescale=1.0e4): + b, channels, length = x.size() + signal = get_timing_signal_1d(length, channels, min_timescale, max_timescale) + return x + signal.to(dtype=x.dtype, device=x.device) + + +def cat_timing_signal_1d(x, min_timescale=1.0, max_timescale=1.0e4, axis=1): + b, channels, length = x.size() + signal = get_timing_signal_1d(length, channels, min_timescale, max_timescale) + return torch.cat([x, signal.to(dtype=x.dtype, device=x.device)], axis) + + +def subsequent_mask(length): + mask = torch.tril(torch.ones(length, length)).unsqueeze(0).unsqueeze(0) + return mask + + +@torch.jit.script +def fused_add_tanh_sigmoid_multiply(input_a, input_b, n_channels): + n_channels_int = n_channels[0] + in_act = input_a + input_b + t_act = torch.tanh(in_act[:, :n_channels_int, :]) + s_act = torch.sigmoid(in_act[:, n_channels_int:, :]) + acts = t_act * s_act + return acts + + +def convert_pad_shape(pad_shape): + l = pad_shape[::-1] + pad_shape = [item for sublist in l for item in sublist] + return pad_shape + + +def shift_1d(x): + x = F.pad(x, convert_pad_shape([[0, 0], [0, 0], [1, 0]]))[:, :, :-1] + return x + + +def sequence_mask(length, max_length=None): + if max_length is None: + max_length = length.max() + x = torch.arange(max_length, dtype=length.dtype, device=length.device) + return x.unsqueeze(0) < length.unsqueeze(1) + + +def generate_path(duration, mask): + """ + duration: [b, 1, t_x] + mask: [b, 1, t_y, t_x] + """ + device = duration.device + + b, _, t_y, t_x = mask.shape + cum_duration = torch.cumsum(duration, -1) + + cum_duration_flat = cum_duration.view(b * t_x) + path = sequence_mask(cum_duration_flat, t_y).to(mask.dtype) + path = path.view(b, t_x, t_y) + path = path - F.pad(path, convert_pad_shape([[0, 0], [1, 0], [0, 0]]))[:, :-1] + path = path.unsqueeze(1).transpose(2,3) * mask + return path + + +def clip_grad_value_(parameters, clip_value, norm_type=2): + if isinstance(parameters, torch.Tensor): + parameters = [parameters] + parameters = list(filter(lambda p: p.grad is not None, parameters)) + norm_type = float(norm_type) + if clip_value is not None: + clip_value = float(clip_value) + + total_norm = 0 + for p in parameters: + param_norm = p.grad.data.norm(norm_type) + total_norm += param_norm.item() ** norm_type + if clip_value is not None: + p.grad.data.clamp_(min=-clip_value, max=clip_value) + total_norm = total_norm ** (1. / norm_type) + return total_norm diff --git a/modules/crepe.py b/modules/crepe.py new file mode 100644 index 0000000000000000000000000000000000000000..0bff0e3474de6483290b56993f9b845e91ef9702 --- /dev/null +++ b/modules/crepe.py @@ -0,0 +1,327 @@ +from typing import Optional,Union +try: + from typing import Literal +except Exception as e: + from typing_extensions import Literal +import numpy as np +import torch +import torchcrepe +from torch import nn +from torch.nn import functional as F +import scipy + +#from:https://github.com/fishaudio/fish-diffusion + +def repeat_expand( + content: Union[torch.Tensor, np.ndarray], target_len: int, mode: str = "nearest" +): + """Repeat content to target length. + This is a wrapper of torch.nn.functional.interpolate. + + Args: + content (torch.Tensor): tensor + target_len (int): target length + mode (str, optional): interpolation mode. Defaults to "nearest". + + Returns: + torch.Tensor: tensor + """ + + ndim = content.ndim + + if content.ndim == 1: + content = content[None, None] + elif content.ndim == 2: + content = content[None] + + assert content.ndim == 3 + + is_np = isinstance(content, np.ndarray) + if is_np: + content = torch.from_numpy(content) + + results = torch.nn.functional.interpolate(content, size=target_len, mode=mode) + + if is_np: + results = results.numpy() + + if ndim == 1: + return results[0, 0] + elif ndim == 2: + return results[0] + + +class BasePitchExtractor: + def __init__( + self, + hop_length: int = 512, + f0_min: float = 50.0, + f0_max: float = 1100.0, + keep_zeros: bool = True, + ): + """Base pitch extractor. + + Args: + hop_length (int, optional): Hop length. Defaults to 512. + f0_min (float, optional): Minimum f0. Defaults to 50.0. + f0_max (float, optional): Maximum f0. Defaults to 1100.0. + keep_zeros (bool, optional): Whether keep zeros in pitch. Defaults to True. + """ + + self.hop_length = hop_length + self.f0_min = f0_min + self.f0_max = f0_max + self.keep_zeros = keep_zeros + + def __call__(self, x, sampling_rate=44100, pad_to=None): + raise NotImplementedError("BasePitchExtractor is not callable.") + + def post_process(self, x, sampling_rate, f0, pad_to): + if isinstance(f0, np.ndarray): + f0 = torch.from_numpy(f0).float().to(x.device) + + if pad_to is None: + return f0 + + f0 = repeat_expand(f0, pad_to) + + if self.keep_zeros: + return f0 + + vuv_vector = torch.zeros_like(f0) + vuv_vector[f0 > 0.0] = 1.0 + vuv_vector[f0 <= 0.0] = 0.0 + + # 去掉0频率, 并线性插值 + nzindex = torch.nonzero(f0).squeeze() + f0 = torch.index_select(f0, dim=0, index=nzindex).cpu().numpy() + time_org = self.hop_length / sampling_rate * nzindex.cpu().numpy() + time_frame = np.arange(pad_to) * self.hop_length / sampling_rate + + if f0.shape[0] <= 0: + return torch.zeros(pad_to, dtype=torch.float, device=x.device),torch.zeros(pad_to, dtype=torch.float, device=x.device) + + if f0.shape[0] == 1: + return torch.ones(pad_to, dtype=torch.float, device=x.device) * f0[0],torch.ones(pad_to, dtype=torch.float, device=x.device) + + # 大概可以用 torch 重写? + f0 = np.interp(time_frame, time_org, f0, left=f0[0], right=f0[-1]) + vuv_vector = vuv_vector.cpu().numpy() + vuv_vector = np.ceil(scipy.ndimage.zoom(vuv_vector,pad_to/len(vuv_vector),order = 0)) + + return f0,vuv_vector + + +class MaskedAvgPool1d(nn.Module): + def __init__( + self, kernel_size: int, stride: Optional[int] = None, padding: Optional[int] = 0 + ): + """An implementation of mean pooling that supports masked values. + + Args: + kernel_size (int): The size of the median pooling window. + stride (int, optional): The stride of the median pooling window. Defaults to None. + padding (int, optional): The padding of the median pooling window. Defaults to 0. + """ + + super(MaskedAvgPool1d, self).__init__() + self.kernel_size = kernel_size + self.stride = stride or kernel_size + self.padding = padding + + def forward(self, x, mask=None): + ndim = x.dim() + if ndim == 2: + x = x.unsqueeze(1) + + assert ( + x.dim() == 3 + ), "Input tensor must have 2 or 3 dimensions (batch_size, channels, width)" + + # Apply the mask by setting masked elements to zero, or make NaNs zero + if mask is None: + mask = ~torch.isnan(x) + + # Ensure mask has the same shape as the input tensor + assert x.shape == mask.shape, "Input tensor and mask must have the same shape" + + masked_x = torch.where(mask, x, torch.zeros_like(x)) + # Create a ones kernel with the same number of channels as the input tensor + ones_kernel = torch.ones(x.size(1), 1, self.kernel_size, device=x.device) + + # Perform sum pooling + sum_pooled = nn.functional.conv1d( + masked_x, + ones_kernel, + stride=self.stride, + padding=self.padding, + groups=x.size(1), + ) + + # Count the non-masked (valid) elements in each pooling window + valid_count = nn.functional.conv1d( + mask.float(), + ones_kernel, + stride=self.stride, + padding=self.padding, + groups=x.size(1), + ) + valid_count = valid_count.clamp(min=1) # Avoid division by zero + + # Perform masked average pooling + avg_pooled = sum_pooled / valid_count + + # Fill zero values with NaNs + avg_pooled[avg_pooled == 0] = float("nan") + + if ndim == 2: + return avg_pooled.squeeze(1) + + return avg_pooled + + +class MaskedMedianPool1d(nn.Module): + def __init__( + self, kernel_size: int, stride: Optional[int] = None, padding: Optional[int] = 0 + ): + """An implementation of median pooling that supports masked values. + + This implementation is inspired by the median pooling implementation in + https://gist.github.com/rwightman/f2d3849281624be7c0f11c85c87c1598 + + Args: + kernel_size (int): The size of the median pooling window. + stride (int, optional): The stride of the median pooling window. Defaults to None. + padding (int, optional): The padding of the median pooling window. Defaults to 0. + """ + + super(MaskedMedianPool1d, self).__init__() + self.kernel_size = kernel_size + self.stride = stride or kernel_size + self.padding = padding + + def forward(self, x, mask=None): + ndim = x.dim() + if ndim == 2: + x = x.unsqueeze(1) + + assert ( + x.dim() == 3 + ), "Input tensor must have 2 or 3 dimensions (batch_size, channels, width)" + + if mask is None: + mask = ~torch.isnan(x) + + assert x.shape == mask.shape, "Input tensor and mask must have the same shape" + + masked_x = torch.where(mask, x, torch.zeros_like(x)) + + x = F.pad(masked_x, (self.padding, self.padding), mode="reflect") + mask = F.pad( + mask.float(), (self.padding, self.padding), mode="constant", value=0 + ) + + x = x.unfold(2, self.kernel_size, self.stride) + mask = mask.unfold(2, self.kernel_size, self.stride) + + x = x.contiguous().view(x.size()[:3] + (-1,)) + mask = mask.contiguous().view(mask.size()[:3] + (-1,)).to(x.device) + + # Combine the mask with the input tensor + #x_masked = torch.where(mask.bool(), x, torch.fill_(torch.zeros_like(x),float("inf"))) + x_masked = torch.where(mask.bool(), x, torch.FloatTensor([float("inf")]).to(x.device)) + + # Sort the masked tensor along the last dimension + x_sorted, _ = torch.sort(x_masked, dim=-1) + + # Compute the count of non-masked (valid) values + valid_count = mask.sum(dim=-1) + + # Calculate the index of the median value for each pooling window + median_idx = (torch.div((valid_count - 1), 2, rounding_mode='trunc')).clamp(min=0) + + # Gather the median values using the calculated indices + median_pooled = x_sorted.gather(-1, median_idx.unsqueeze(-1).long()).squeeze(-1) + + # Fill infinite values with NaNs + median_pooled[torch.isinf(median_pooled)] = float("nan") + + if ndim == 2: + return median_pooled.squeeze(1) + + return median_pooled + + +class CrepePitchExtractor(BasePitchExtractor): + def __init__( + self, + hop_length: int = 512, + f0_min: float = 50.0, + f0_max: float = 1100.0, + threshold: float = 0.05, + keep_zeros: bool = False, + device = None, + model: Literal["full", "tiny"] = "full", + use_fast_filters: bool = True, + ): + super().__init__(hop_length, f0_min, f0_max, keep_zeros) + + self.threshold = threshold + self.model = model + self.use_fast_filters = use_fast_filters + self.hop_length = hop_length + if device is None: + self.dev = torch.device("cuda" if torch.cuda.is_available() else "cpu") + else: + self.dev = torch.device(device) + if self.use_fast_filters: + self.median_filter = MaskedMedianPool1d(3, 1, 1).to(device) + self.mean_filter = MaskedAvgPool1d(3, 1, 1).to(device) + + def __call__(self, x, sampling_rate=44100, pad_to=None): + """Extract pitch using crepe. + + + Args: + x (torch.Tensor): Audio signal, shape (1, T). + sampling_rate (int, optional): Sampling rate. Defaults to 44100. + pad_to (int, optional): Pad to length. Defaults to None. + + Returns: + torch.Tensor: Pitch, shape (T // hop_length,). + """ + + assert x.ndim == 2, f"Expected 2D tensor, got {x.ndim}D tensor." + assert x.shape[0] == 1, f"Expected 1 channel, got {x.shape[0]} channels." + + x = x.to(self.dev) + f0, pd = torchcrepe.predict( + x, + sampling_rate, + self.hop_length, + self.f0_min, + self.f0_max, + pad=True, + model=self.model, + batch_size=1024, + device=x.device, + return_periodicity=True, + ) + + # Filter, remove silence, set uv threshold, refer to the original warehouse readme + if self.use_fast_filters: + pd = self.median_filter(pd) + else: + pd = torchcrepe.filter.median(pd, 3) + + pd = torchcrepe.threshold.Silence(-60.0)(pd, x, sampling_rate, 512) + f0 = torchcrepe.threshold.At(self.threshold)(f0, pd) + + if self.use_fast_filters: + f0 = self.mean_filter(f0) + else: + f0 = torchcrepe.filter.mean(f0, 3) + + f0 = torch.where(torch.isnan(f0), torch.full_like(f0, 0), f0)[0] + + return self.post_process(x, sampling_rate, f0, pad_to) diff --git a/modules/enhancer.py b/modules/enhancer.py new file mode 100644 index 0000000000000000000000000000000000000000..37676311f7d8dc4ddc2a5244dedc27b2437e04f5 --- /dev/null +++ b/modules/enhancer.py @@ -0,0 +1,105 @@ +import numpy as np +import torch +import torch.nn.functional as F +from vdecoder.nsf_hifigan.nvSTFT import STFT +from vdecoder.nsf_hifigan.models import load_model +from torchaudio.transforms import Resample + +class Enhancer: + def __init__(self, enhancer_type, enhancer_ckpt, device=None): + if device is None: + device = 'cuda' if torch.cuda.is_available() else 'cpu' + self.device = device + + if enhancer_type == 'nsf-hifigan': + self.enhancer = NsfHifiGAN(enhancer_ckpt, device=self.device) + else: + raise ValueError(f" [x] Unknown enhancer: {enhancer_type}") + + self.resample_kernel = {} + self.enhancer_sample_rate = self.enhancer.sample_rate() + self.enhancer_hop_size = self.enhancer.hop_size() + + def enhance(self, + audio, # 1, T + sample_rate, + f0, # 1, n_frames, 1 + hop_size, + adaptive_key = 0, + silence_front = 0 + ): + # enhancer start time + start_frame = int(silence_front * sample_rate / hop_size) + real_silence_front = start_frame * hop_size / sample_rate + audio = audio[:, int(np.round(real_silence_front * sample_rate)) : ] + f0 = f0[: , start_frame :, :] + + # adaptive parameters + adaptive_factor = 2 ** ( -adaptive_key / 12) + adaptive_sample_rate = 100 * int(np.round(self.enhancer_sample_rate / adaptive_factor / 100)) + real_factor = self.enhancer_sample_rate / adaptive_sample_rate + + # resample the ddsp output + if sample_rate == adaptive_sample_rate: + audio_res = audio + else: + key_str = str(sample_rate) + str(adaptive_sample_rate) + if key_str not in self.resample_kernel: + self.resample_kernel[key_str] = Resample(sample_rate, adaptive_sample_rate, lowpass_filter_width = 128).to(self.device) + audio_res = self.resample_kernel[key_str](audio) + + n_frames = int(audio_res.size(-1) // self.enhancer_hop_size + 1) + + # resample f0 + f0_np = f0.squeeze(0).squeeze(-1).cpu().numpy() + f0_np *= real_factor + time_org = (hop_size / sample_rate) * np.arange(len(f0_np)) / real_factor + time_frame = (self.enhancer_hop_size / self.enhancer_sample_rate) * np.arange(n_frames) + f0_res = np.interp(time_frame, time_org, f0_np, left=f0_np[0], right=f0_np[-1]) + f0_res = torch.from_numpy(f0_res).unsqueeze(0).float().to(self.device) # 1, n_frames + + # enhance + enhanced_audio, enhancer_sample_rate = self.enhancer(audio_res, f0_res) + + # resample the enhanced output + if adaptive_factor != 0: + key_str = str(adaptive_sample_rate) + str(enhancer_sample_rate) + if key_str not in self.resample_kernel: + self.resample_kernel[key_str] = Resample(adaptive_sample_rate, enhancer_sample_rate, lowpass_filter_width = 128).to(self.device) + enhanced_audio = self.resample_kernel[key_str](enhanced_audio) + + # pad the silence frames + if start_frame > 0: + enhanced_audio = F.pad(enhanced_audio, (int(np.round(enhancer_sample_rate * real_silence_front)), 0)) + + return enhanced_audio, enhancer_sample_rate + + +class NsfHifiGAN(torch.nn.Module): + def __init__(self, model_path, device=None): + super().__init__() + if device is None: + device = 'cuda' if torch.cuda.is_available() else 'cpu' + self.device = device + print('| Load HifiGAN: ', model_path) + self.model, self.h = load_model(model_path, device=self.device) + + def sample_rate(self): + return self.h.sampling_rate + + def hop_size(self): + return self.h.hop_size + + def forward(self, audio, f0): + stft = STFT( + self.h.sampling_rate, + self.h.num_mels, + self.h.n_fft, + self.h.win_size, + self.h.hop_size, + self.h.fmin, + self.h.fmax) + with torch.no_grad(): + mel = stft.get_mel(audio) + enhanced_audio = self.model(mel, f0[:,:mel.size(-1)]).view(-1) + return enhanced_audio, self.h.sampling_rate \ No newline at end of file diff --git a/modules/losses.py b/modules/losses.py new file mode 100644 index 0000000000000000000000000000000000000000..cd21799eccde350c3aac0bdd661baf96ed220147 --- /dev/null +++ b/modules/losses.py @@ -0,0 +1,61 @@ +import torch +from torch.nn import functional as F + +import modules.commons as commons + + +def feature_loss(fmap_r, fmap_g): + loss = 0 + for dr, dg in zip(fmap_r, fmap_g): + for rl, gl in zip(dr, dg): + rl = rl.float().detach() + gl = gl.float() + loss += torch.mean(torch.abs(rl - gl)) + + return loss * 2 + + +def discriminator_loss(disc_real_outputs, disc_generated_outputs): + loss = 0 + r_losses = [] + g_losses = [] + for dr, dg in zip(disc_real_outputs, disc_generated_outputs): + dr = dr.float() + dg = dg.float() + r_loss = torch.mean((1-dr)**2) + g_loss = torch.mean(dg**2) + loss += (r_loss + g_loss) + r_losses.append(r_loss.item()) + g_losses.append(g_loss.item()) + + return loss, r_losses, g_losses + + +def generator_loss(disc_outputs): + loss = 0 + gen_losses = [] + for dg in disc_outputs: + dg = dg.float() + l = torch.mean((1-dg)**2) + gen_losses.append(l) + loss += l + + return loss, gen_losses + + +def kl_loss(z_p, logs_q, m_p, logs_p, z_mask): + """ + z_p, logs_q: [b, h, t_t] + m_p, logs_p: [b, h, t_t] + """ + z_p = z_p.float() + logs_q = logs_q.float() + m_p = m_p.float() + logs_p = logs_p.float() + z_mask = z_mask.float() + #print(logs_p) + kl = logs_p - logs_q - 0.5 + kl += 0.5 * ((z_p - m_p)**2) * torch.exp(-2. * logs_p) + kl = torch.sum(kl * z_mask) + l = kl / torch.sum(z_mask) + return l diff --git a/modules/mel_processing.py b/modules/mel_processing.py new file mode 100644 index 0000000000000000000000000000000000000000..99c5b35beb83f3b288af0fac5b49ebf2c69f062c --- /dev/null +++ b/modules/mel_processing.py @@ -0,0 +1,112 @@ +import math +import os +import random +import torch +from torch import nn +import torch.nn.functional as F +import torch.utils.data +import numpy as np +import librosa +import librosa.util as librosa_util +from librosa.util import normalize, pad_center, tiny +from scipy.signal import get_window +from scipy.io.wavfile import read +from librosa.filters import mel as librosa_mel_fn + +MAX_WAV_VALUE = 32768.0 + + +def dynamic_range_compression_torch(x, C=1, clip_val=1e-5): + """ + PARAMS + ------ + C: compression factor + """ + return torch.log(torch.clamp(x, min=clip_val) * C) + + +def dynamic_range_decompression_torch(x, C=1): + """ + PARAMS + ------ + C: compression factor used to compress + """ + return torch.exp(x) / C + + +def spectral_normalize_torch(magnitudes): + output = dynamic_range_compression_torch(magnitudes) + return output + + +def spectral_de_normalize_torch(magnitudes): + output = dynamic_range_decompression_torch(magnitudes) + return output + + +mel_basis = {} +hann_window = {} + + +def spectrogram_torch(y, n_fft, sampling_rate, hop_size, win_size, center=False): + if torch.min(y) < -1.: + print('min value is ', torch.min(y)) + if torch.max(y) > 1.: + print('max value is ', torch.max(y)) + + global hann_window + dtype_device = str(y.dtype) + '_' + str(y.device) + wnsize_dtype_device = str(win_size) + '_' + dtype_device + if wnsize_dtype_device not in hann_window: + hann_window[wnsize_dtype_device] = torch.hann_window(win_size).to(dtype=y.dtype, device=y.device) + + y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_size)/2), int((n_fft-hop_size)/2)), mode='reflect') + y = y.squeeze(1) + + spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[wnsize_dtype_device], + center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=False) + + spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6) + return spec + + +def spec_to_mel_torch(spec, n_fft, num_mels, sampling_rate, fmin, fmax): + global mel_basis + dtype_device = str(spec.dtype) + '_' + str(spec.device) + fmax_dtype_device = str(fmax) + '_' + dtype_device + if fmax_dtype_device not in mel_basis: + mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax) + mel_basis[fmax_dtype_device] = torch.from_numpy(mel).to(dtype=spec.dtype, device=spec.device) + spec = torch.matmul(mel_basis[fmax_dtype_device], spec) + spec = spectral_normalize_torch(spec) + return spec + + +def mel_spectrogram_torch(y, n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax, center=False): + if torch.min(y) < -1.: + print('min value is ', torch.min(y)) + if torch.max(y) > 1.: + print('max value is ', torch.max(y)) + + global mel_basis, hann_window + dtype_device = str(y.dtype) + '_' + str(y.device) + fmax_dtype_device = str(fmax) + '_' + dtype_device + wnsize_dtype_device = str(win_size) + '_' + dtype_device + if fmax_dtype_device not in mel_basis: + mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax) + mel_basis[fmax_dtype_device] = torch.from_numpy(mel).to(dtype=y.dtype, device=y.device) + if wnsize_dtype_device not in hann_window: + hann_window[wnsize_dtype_device] = torch.hann_window(win_size).to(dtype=y.dtype, device=y.device) + + y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_size)/2), int((n_fft-hop_size)/2)), mode='reflect') + y = y.squeeze(1) + + spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[wnsize_dtype_device], + center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=False) + + spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6) + + spec = torch.matmul(mel_basis[fmax_dtype_device], spec) + spec = spectral_normalize_torch(spec) + + return spec diff --git a/modules/modules.py b/modules/modules.py new file mode 100644 index 0000000000000000000000000000000000000000..54290fd207b25e93831bd21005990ea137e6b50e --- /dev/null +++ b/modules/modules.py @@ -0,0 +1,342 @@ +import copy +import math +import numpy as np +import scipy +import torch +from torch import nn +from torch.nn import functional as F + +from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d +from torch.nn.utils import weight_norm, remove_weight_norm + +import modules.commons as commons +from modules.commons import init_weights, get_padding + + +LRELU_SLOPE = 0.1 + + +class LayerNorm(nn.Module): + def __init__(self, channels, eps=1e-5): + super().__init__() + self.channels = channels + self.eps = eps + + self.gamma = nn.Parameter(torch.ones(channels)) + self.beta = nn.Parameter(torch.zeros(channels)) + + def forward(self, x): + x = x.transpose(1, -1) + x = F.layer_norm(x, (self.channels,), self.gamma, self.beta, self.eps) + return x.transpose(1, -1) + + +class ConvReluNorm(nn.Module): + def __init__(self, in_channels, hidden_channels, out_channels, kernel_size, n_layers, p_dropout): + super().__init__() + self.in_channels = in_channels + self.hidden_channels = hidden_channels + self.out_channels = out_channels + self.kernel_size = kernel_size + self.n_layers = n_layers + self.p_dropout = p_dropout + assert n_layers > 1, "Number of layers should be larger than 0." + + self.conv_layers = nn.ModuleList() + self.norm_layers = nn.ModuleList() + self.conv_layers.append(nn.Conv1d(in_channels, hidden_channels, kernel_size, padding=kernel_size//2)) + self.norm_layers.append(LayerNorm(hidden_channels)) + self.relu_drop = nn.Sequential( + nn.ReLU(), + nn.Dropout(p_dropout)) + for _ in range(n_layers-1): + self.conv_layers.append(nn.Conv1d(hidden_channels, hidden_channels, kernel_size, padding=kernel_size//2)) + self.norm_layers.append(LayerNorm(hidden_channels)) + self.proj = nn.Conv1d(hidden_channels, out_channels, 1) + self.proj.weight.data.zero_() + self.proj.bias.data.zero_() + + def forward(self, x, x_mask): + x_org = x + for i in range(self.n_layers): + x = self.conv_layers[i](x * x_mask) + x = self.norm_layers[i](x) + x = self.relu_drop(x) + x = x_org + self.proj(x) + return x * x_mask + + +class DDSConv(nn.Module): + """ + Dialted and Depth-Separable Convolution + """ + def __init__(self, channels, kernel_size, n_layers, p_dropout=0.): + super().__init__() + self.channels = channels + self.kernel_size = kernel_size + self.n_layers = n_layers + self.p_dropout = p_dropout + + self.drop = nn.Dropout(p_dropout) + self.convs_sep = nn.ModuleList() + self.convs_1x1 = nn.ModuleList() + self.norms_1 = nn.ModuleList() + self.norms_2 = nn.ModuleList() + for i in range(n_layers): + dilation = kernel_size ** i + padding = (kernel_size * dilation - dilation) // 2 + self.convs_sep.append(nn.Conv1d(channels, channels, kernel_size, + groups=channels, dilation=dilation, padding=padding + )) + self.convs_1x1.append(nn.Conv1d(channels, channels, 1)) + self.norms_1.append(LayerNorm(channels)) + self.norms_2.append(LayerNorm(channels)) + + def forward(self, x, x_mask, g=None): + if g is not None: + x = x + g + for i in range(self.n_layers): + y = self.convs_sep[i](x * x_mask) + y = self.norms_1[i](y) + y = F.gelu(y) + y = self.convs_1x1[i](y) + y = self.norms_2[i](y) + y = F.gelu(y) + y = self.drop(y) + x = x + y + return x * x_mask + + +class WN(torch.nn.Module): + def __init__(self, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=0, p_dropout=0): + super(WN, self).__init__() + assert(kernel_size % 2 == 1) + self.hidden_channels =hidden_channels + self.kernel_size = kernel_size, + self.dilation_rate = dilation_rate + self.n_layers = n_layers + self.gin_channels = gin_channels + self.p_dropout = p_dropout + + self.in_layers = torch.nn.ModuleList() + self.res_skip_layers = torch.nn.ModuleList() + self.drop = nn.Dropout(p_dropout) + + if gin_channels != 0: + cond_layer = torch.nn.Conv1d(gin_channels, 2*hidden_channels*n_layers, 1) + self.cond_layer = torch.nn.utils.weight_norm(cond_layer, name='weight') + + for i in range(n_layers): + dilation = dilation_rate ** i + padding = int((kernel_size * dilation - dilation) / 2) + in_layer = torch.nn.Conv1d(hidden_channels, 2*hidden_channels, kernel_size, + dilation=dilation, padding=padding) + in_layer = torch.nn.utils.weight_norm(in_layer, name='weight') + self.in_layers.append(in_layer) + + # last one is not necessary + if i < n_layers - 1: + res_skip_channels = 2 * hidden_channels + else: + res_skip_channels = hidden_channels + + res_skip_layer = torch.nn.Conv1d(hidden_channels, res_skip_channels, 1) + res_skip_layer = torch.nn.utils.weight_norm(res_skip_layer, name='weight') + self.res_skip_layers.append(res_skip_layer) + + def forward(self, x, x_mask, g=None, **kwargs): + output = torch.zeros_like(x) + n_channels_tensor = torch.IntTensor([self.hidden_channels]) + + if g is not None: + g = self.cond_layer(g) + + for i in range(self.n_layers): + x_in = self.in_layers[i](x) + if g is not None: + cond_offset = i * 2 * self.hidden_channels + g_l = g[:,cond_offset:cond_offset+2*self.hidden_channels,:] + else: + g_l = torch.zeros_like(x_in) + + acts = commons.fused_add_tanh_sigmoid_multiply( + x_in, + g_l, + n_channels_tensor) + acts = self.drop(acts) + + res_skip_acts = self.res_skip_layers[i](acts) + if i < self.n_layers - 1: + res_acts = res_skip_acts[:,:self.hidden_channels,:] + x = (x + res_acts) * x_mask + output = output + res_skip_acts[:,self.hidden_channels:,:] + else: + output = output + res_skip_acts + return output * x_mask + + def remove_weight_norm(self): + if self.gin_channels != 0: + torch.nn.utils.remove_weight_norm(self.cond_layer) + for l in self.in_layers: + torch.nn.utils.remove_weight_norm(l) + for l in self.res_skip_layers: + torch.nn.utils.remove_weight_norm(l) + + +class ResBlock1(torch.nn.Module): + def __init__(self, channels, kernel_size=3, dilation=(1, 3, 5)): + super(ResBlock1, self).__init__() + self.convs1 = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0], + padding=get_padding(kernel_size, dilation[0]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1], + padding=get_padding(kernel_size, dilation[1]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[2], + padding=get_padding(kernel_size, dilation[2]))) + ]) + self.convs1.apply(init_weights) + + self.convs2 = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))) + ]) + self.convs2.apply(init_weights) + + def forward(self, x, x_mask=None): + for c1, c2 in zip(self.convs1, self.convs2): + xt = F.leaky_relu(x, LRELU_SLOPE) + if x_mask is not None: + xt = xt * x_mask + xt = c1(xt) + xt = F.leaky_relu(xt, LRELU_SLOPE) + if x_mask is not None: + xt = xt * x_mask + xt = c2(xt) + x = xt + x + if x_mask is not None: + x = x * x_mask + return x + + def remove_weight_norm(self): + for l in self.convs1: + remove_weight_norm(l) + for l in self.convs2: + remove_weight_norm(l) + + +class ResBlock2(torch.nn.Module): + def __init__(self, channels, kernel_size=3, dilation=(1, 3)): + super(ResBlock2, self).__init__() + self.convs = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0], + padding=get_padding(kernel_size, dilation[0]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1], + padding=get_padding(kernel_size, dilation[1]))) + ]) + self.convs.apply(init_weights) + + def forward(self, x, x_mask=None): + for c in self.convs: + xt = F.leaky_relu(x, LRELU_SLOPE) + if x_mask is not None: + xt = xt * x_mask + xt = c(xt) + x = xt + x + if x_mask is not None: + x = x * x_mask + return x + + def remove_weight_norm(self): + for l in self.convs: + remove_weight_norm(l) + + +class Log(nn.Module): + def forward(self, x, x_mask, reverse=False, **kwargs): + if not reverse: + y = torch.log(torch.clamp_min(x, 1e-5)) * x_mask + logdet = torch.sum(-y, [1, 2]) + return y, logdet + else: + x = torch.exp(x) * x_mask + return x + + +class Flip(nn.Module): + def forward(self, x, *args, reverse=False, **kwargs): + x = torch.flip(x, [1]) + if not reverse: + logdet = torch.zeros(x.size(0)).to(dtype=x.dtype, device=x.device) + return x, logdet + else: + return x + + +class ElementwiseAffine(nn.Module): + def __init__(self, channels): + super().__init__() + self.channels = channels + self.m = nn.Parameter(torch.zeros(channels,1)) + self.logs = nn.Parameter(torch.zeros(channels,1)) + + def forward(self, x, x_mask, reverse=False, **kwargs): + if not reverse: + y = self.m + torch.exp(self.logs) * x + y = y * x_mask + logdet = torch.sum(self.logs * x_mask, [1,2]) + return y, logdet + else: + x = (x - self.m) * torch.exp(-self.logs) * x_mask + return x + + +class ResidualCouplingLayer(nn.Module): + def __init__(self, + channels, + hidden_channels, + kernel_size, + dilation_rate, + n_layers, + p_dropout=0, + gin_channels=0, + mean_only=False): + assert channels % 2 == 0, "channels should be divisible by 2" + super().__init__() + self.channels = channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.dilation_rate = dilation_rate + self.n_layers = n_layers + self.half_channels = channels // 2 + self.mean_only = mean_only + + self.pre = nn.Conv1d(self.half_channels, hidden_channels, 1) + self.enc = WN(hidden_channels, kernel_size, dilation_rate, n_layers, p_dropout=p_dropout, gin_channels=gin_channels) + self.post = nn.Conv1d(hidden_channels, self.half_channels * (2 - mean_only), 1) + self.post.weight.data.zero_() + self.post.bias.data.zero_() + + def forward(self, x, x_mask, g=None, reverse=False): + x0, x1 = torch.split(x, [self.half_channels]*2, 1) + h = self.pre(x0) * x_mask + h = self.enc(h, x_mask, g=g) + stats = self.post(h) * x_mask + if not self.mean_only: + m, logs = torch.split(stats, [self.half_channels]*2, 1) + else: + m = stats + logs = torch.zeros_like(m) + + if not reverse: + x1 = m + x1 * torch.exp(logs) * x_mask + x = torch.cat([x0, x1], 1) + logdet = torch.sum(logs, [1,2]) + return x, logdet + else: + x1 = (x1 - m) * torch.exp(-logs) * x_mask + x = torch.cat([x0, x1], 1) + return x diff --git a/onnx_export.py b/onnx_export.py new file mode 100644 index 0000000000000000000000000000000000000000..a70a912cc1b6dd908ff6496bbc6fa8dd576e233b --- /dev/null +++ b/onnx_export.py @@ -0,0 +1,54 @@ +import torch +from onnxexport.model_onnx import SynthesizerTrn +import utils + +def main(NetExport): + path = "SoVits4.0" + if NetExport: + device = torch.device("cpu") + hps = utils.get_hparams_from_file(f"checkpoints/{path}/config.json") + SVCVITS = SynthesizerTrn( + hps.data.filter_length // 2 + 1, + hps.train.segment_size // hps.data.hop_length, + **hps.model) + _ = utils.load_checkpoint(f"checkpoints/{path}/model.pth", SVCVITS, None) + _ = SVCVITS.eval().to(device) + for i in SVCVITS.parameters(): + i.requires_grad = False + + n_frame = 10 + test_hidden_unit = torch.rand(1, n_frame, 256) + test_pitch = torch.rand(1, n_frame) + test_mel2ph = torch.arange(0, n_frame, dtype=torch.int64)[None] # torch.LongTensor([0, 1, 2, 3, 4, 5, 6, 7, 8, 9]).unsqueeze(0) + test_uv = torch.ones(1, n_frame, dtype=torch.float32) + test_noise = torch.randn(1, 192, n_frame) + test_sid = torch.LongTensor([0]) + input_names = ["c", "f0", "mel2ph", "uv", "noise", "sid"] + output_names = ["audio", ] + + torch.onnx.export(SVCVITS, + ( + test_hidden_unit.to(device), + test_pitch.to(device), + test_mel2ph.to(device), + test_uv.to(device), + test_noise.to(device), + test_sid.to(device) + ), + f"checkpoints/{path}/model.onnx", + dynamic_axes={ + "c": [0, 1], + "f0": [1], + "mel2ph": [1], + "uv": [1], + "noise": [2], + }, + do_constant_folding=False, + opset_version=16, + verbose=False, + input_names=input_names, + output_names=output_names) + + +if __name__ == '__main__': + main(True) diff --git a/onnxexport/model_onnx.py b/onnxexport/model_onnx.py new file mode 100644 index 0000000000000000000000000000000000000000..e28bae95ec1e53aa05d06fc784ff86d55f228d60 --- /dev/null +++ b/onnxexport/model_onnx.py @@ -0,0 +1,335 @@ +import torch +from torch import nn +from torch.nn import functional as F + +import modules.attentions as attentions +import modules.commons as commons +import modules.modules as modules + +from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d +from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm + +import utils +from modules.commons import init_weights, get_padding +from vdecoder.hifigan.models import Generator +from utils import f0_to_coarse + + +class ResidualCouplingBlock(nn.Module): + def __init__(self, + channels, + hidden_channels, + kernel_size, + dilation_rate, + n_layers, + n_flows=4, + gin_channels=0): + super().__init__() + self.channels = channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.dilation_rate = dilation_rate + self.n_layers = n_layers + self.n_flows = n_flows + self.gin_channels = gin_channels + + self.flows = nn.ModuleList() + for i in range(n_flows): + self.flows.append( + modules.ResidualCouplingLayer(channels, hidden_channels, kernel_size, dilation_rate, n_layers, + gin_channels=gin_channels, mean_only=True)) + self.flows.append(modules.Flip()) + + def forward(self, x, x_mask, g=None, reverse=False): + if not reverse: + for flow in self.flows: + x, _ = flow(x, x_mask, g=g, reverse=reverse) + else: + for flow in reversed(self.flows): + x = flow(x, x_mask, g=g, reverse=reverse) + return x + + +class Encoder(nn.Module): + def __init__(self, + in_channels, + out_channels, + hidden_channels, + kernel_size, + dilation_rate, + n_layers, + gin_channels=0): + super().__init__() + self.in_channels = in_channels + self.out_channels = out_channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.dilation_rate = dilation_rate + self.n_layers = n_layers + self.gin_channels = gin_channels + + self.pre = nn.Conv1d(in_channels, hidden_channels, 1) + self.enc = modules.WN(hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels) + self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1) + + def forward(self, x, x_lengths, g=None): + # print(x.shape,x_lengths.shape) + x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype) + x = self.pre(x) * x_mask + x = self.enc(x, x_mask, g=g) + stats = self.proj(x) * x_mask + m, logs = torch.split(stats, self.out_channels, dim=1) + z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask + return z, m, logs, x_mask + + +class TextEncoder(nn.Module): + def __init__(self, + out_channels, + hidden_channels, + kernel_size, + n_layers, + gin_channels=0, + filter_channels=None, + n_heads=None, + p_dropout=None): + super().__init__() + self.out_channels = out_channels + self.hidden_channels = hidden_channels + self.kernel_size = kernel_size + self.n_layers = n_layers + self.gin_channels = gin_channels + self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1) + self.f0_emb = nn.Embedding(256, hidden_channels) + + self.enc_ = attentions.Encoder( + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout) + + def forward(self, x, x_mask, f0=None, z=None): + x = x + self.f0_emb(f0).transpose(1, 2) + x = self.enc_(x * x_mask, x_mask) + stats = self.proj(x) * x_mask + m, logs = torch.split(stats, self.out_channels, dim=1) + z = (m + z * torch.exp(logs)) * x_mask + return z, m, logs, x_mask + + +class DiscriminatorP(torch.nn.Module): + def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False): + super(DiscriminatorP, self).__init__() + self.period = period + self.use_spectral_norm = use_spectral_norm + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))), + norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(get_padding(kernel_size, 1), 0))), + ]) + self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0))) + + def forward(self, x): + fmap = [] + + # 1d to 2d + b, c, t = x.shape + if t % self.period != 0: # pad first + n_pad = self.period - (t % self.period) + x = F.pad(x, (0, n_pad), "reflect") + t = t + n_pad + x = x.view(b, c, t // self.period, self.period) + + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, modules.LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class DiscriminatorS(torch.nn.Module): + def __init__(self, use_spectral_norm=False): + super(DiscriminatorS, self).__init__() + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv1d(1, 16, 15, 1, padding=7)), + norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)), + norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)), + norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)), + norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)), + norm_f(Conv1d(1024, 1024, 5, 1, padding=2)), + ]) + self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1)) + + def forward(self, x): + fmap = [] + + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, modules.LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class F0Decoder(nn.Module): + def __init__(self, + out_channels, + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout, + spk_channels=0): + super().__init__() + self.out_channels = out_channels + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.spk_channels = spk_channels + + self.prenet = nn.Conv1d(hidden_channels, hidden_channels, 3, padding=1) + self.decoder = attentions.FFT( + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout) + self.proj = nn.Conv1d(hidden_channels, out_channels, 1) + self.f0_prenet = nn.Conv1d(1, hidden_channels, 3, padding=1) + self.cond = nn.Conv1d(spk_channels, hidden_channels, 1) + + def forward(self, x, norm_f0, x_mask, spk_emb=None): + x = torch.detach(x) + if spk_emb is not None: + x = x + self.cond(spk_emb) + x += self.f0_prenet(norm_f0) + x = self.prenet(x) * x_mask + x = self.decoder(x * x_mask, x_mask) + x = self.proj(x) * x_mask + return x + + +class SynthesizerTrn(nn.Module): + """ + Synthesizer for Training + """ + + def __init__(self, + spec_channels, + segment_size, + inter_channels, + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout, + resblock, + resblock_kernel_sizes, + resblock_dilation_sizes, + upsample_rates, + upsample_initial_channel, + upsample_kernel_sizes, + gin_channels, + ssl_dim, + n_speakers, + sampling_rate=44100, + **kwargs): + super().__init__() + self.spec_channels = spec_channels + self.inter_channels = inter_channels + self.hidden_channels = hidden_channels + self.filter_channels = filter_channels + self.n_heads = n_heads + self.n_layers = n_layers + self.kernel_size = kernel_size + self.p_dropout = p_dropout + self.resblock = resblock + self.resblock_kernel_sizes = resblock_kernel_sizes + self.resblock_dilation_sizes = resblock_dilation_sizes + self.upsample_rates = upsample_rates + self.upsample_initial_channel = upsample_initial_channel + self.upsample_kernel_sizes = upsample_kernel_sizes + self.segment_size = segment_size + self.gin_channels = gin_channels + self.ssl_dim = ssl_dim + self.emb_g = nn.Embedding(n_speakers, gin_channels) + + self.pre = nn.Conv1d(ssl_dim, hidden_channels, kernel_size=5, padding=2) + + self.enc_p = TextEncoder( + inter_channels, + hidden_channels, + filter_channels=filter_channels, + n_heads=n_heads, + n_layers=n_layers, + kernel_size=kernel_size, + p_dropout=p_dropout + ) + hps = { + "sampling_rate": sampling_rate, + "inter_channels": inter_channels, + "resblock": resblock, + "resblock_kernel_sizes": resblock_kernel_sizes, + "resblock_dilation_sizes": resblock_dilation_sizes, + "upsample_rates": upsample_rates, + "upsample_initial_channel": upsample_initial_channel, + "upsample_kernel_sizes": upsample_kernel_sizes, + "gin_channels": gin_channels, + } + self.dec = Generator(h=hps) + self.enc_q = Encoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16, gin_channels=gin_channels) + self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels) + self.f0_decoder = F0Decoder( + 1, + hidden_channels, + filter_channels, + n_heads, + n_layers, + kernel_size, + p_dropout, + spk_channels=gin_channels + ) + self.emb_uv = nn.Embedding(2, hidden_channels) + self.predict_f0 = False + + def forward(self, c, f0, mel2ph, uv, noise=None, g=None): + + decoder_inp = F.pad(c, [0, 0, 1, 0]) + mel2ph_ = mel2ph.unsqueeze(2).repeat([1, 1, c.shape[-1]]) + c = torch.gather(decoder_inp, 1, mel2ph_).transpose(1, 2) # [B, T, H] + + c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device) + g = g.unsqueeze(0) + g = self.emb_g(g).transpose(1, 2) + x_mask = torch.unsqueeze(commons.sequence_mask(c_lengths, c.size(2)), 1).to(c.dtype) + x = self.pre(c) * x_mask + self.emb_uv(uv.long()).transpose(1, 2) + + if self.predict_f0: + lf0 = 2595. * torch.log10(1. + f0.unsqueeze(1) / 700.) / 500 + norm_lf0 = utils.normalize_f0(lf0, x_mask, uv, random_scale=False) + pred_lf0 = self.f0_decoder(x, norm_lf0, x_mask, spk_emb=g) + f0 = (700 * (torch.pow(10, pred_lf0 * 500 / 2595) - 1)).squeeze(1) + + z_p, m_p, logs_p, c_mask = self.enc_p(x, x_mask, f0=f0_to_coarse(f0), z=noise) + z = self.flow(z_p, c_mask, g=g, reverse=True) + o = self.dec(z * c_mask, g=g, f0=f0) + return o diff --git a/preprocess_flist_config.py b/preprocess_flist_config.py new file mode 100644 index 0000000000000000000000000000000000000000..2717e5132644c596e14dc04bd4d35235fdec058e --- /dev/null +++ b/preprocess_flist_config.py @@ -0,0 +1,75 @@ +import os +import argparse +import re + +from tqdm import tqdm +from random import shuffle +import json +import wave + +config_template = json.load(open("configs_template/config_template.json")) + +pattern = re.compile(r'^[\.a-zA-Z0-9_\/]+$') + +def get_wav_duration(file_path): + with wave.open(file_path, 'rb') as wav_file: + # 获取音频帧数 + n_frames = wav_file.getnframes() + # 获取采样率 + framerate = wav_file.getframerate() + # 计算时长（秒） + duration = n_frames / float(framerate) + return duration + +if __name__ == "__main__": + parser = argparse.ArgumentParser() + parser.add_argument("--train_list", type=str, default="./filelists/train.txt", help="path to train list") + parser.add_argument("--val_list", type=str, default="./filelists/val.txt", help="path to val list") + parser.add_argument("--source_dir", type=str, default="./dataset/44k", help="path to source dir") + args = parser.parse_args() + + train = [] + val = [] + idx = 0 + spk_dict = {} + spk_id = 0 + for speaker in tqdm(os.listdir(args.source_dir)): + spk_dict[speaker] = spk_id + spk_id += 1 + wavs = ["/".join([args.source_dir, speaker, i]) for i in os.listdir(os.path.join(args.source_dir, speaker))] + new_wavs = [] + for file in wavs: + if not file.endswith("wav"): + continue + if not pattern.match(file): + print(f"warning：文件名{file}中包含非字母数字下划线，可能会导致错误。（也可能不会）") + if get_wav_duration(file) < 0.3: + print("skip too short audio:", file) + continue + new_wavs.append(file) + wavs = new_wavs + shuffle(wavs) + train += wavs[2:] + val += wavs[:2] + + shuffle(train) + shuffle(val) + + print("Writing", args.train_list) + with open(args.train_list, "w") as f: + for fname in tqdm(train): + wavpath = fname + f.write(wavpath + "\n") + + print("Writing", args.val_list) + with open(args.val_list, "w") as f: + for fname in tqdm(val): + wavpath = fname + f.write(wavpath + "\n") + + config_template["spk"] = spk_dict + config_template["model"]["n_speakers"] = spk_id + + print("Writing configs/config.json") + with open("configs/config.json", "w") as f: + json.dump(config_template, f, indent=2) diff --git a/preprocess_hubert_f0.py b/preprocess_hubert_f0.py new file mode 100644 index 0000000000000000000000000000000000000000..763fb0d65540ed4d62b269914e81c740f3ff6bba --- /dev/null +++ b/preprocess_hubert_f0.py @@ -0,0 +1,101 @@ +import math +import multiprocessing +import os +import argparse +from random import shuffle + +import torch +from glob import glob +from tqdm import tqdm +from modules.mel_processing import spectrogram_torch + +import utils +import logging + +logging.getLogger("numba").setLevel(logging.WARNING) +import librosa +import numpy as np + +hps = utils.get_hparams_from_file("configs/config.json") +sampling_rate = hps.data.sampling_rate +hop_length = hps.data.hop_length + + +def process_one(filename, hmodel): + # print(filename) + wav, sr = librosa.load(filename, sr=sampling_rate) + soft_path = filename + ".soft.pt" + if not os.path.exists(soft_path): + device = torch.device("cuda" if torch.cuda.is_available() else "cpu") + wav16k = librosa.resample(wav, orig_sr=sampling_rate, target_sr=16000) + wav16k = torch.from_numpy(wav16k).to(device) + c = utils.get_hubert_content(hmodel, wav_16k_tensor=wav16k) + torch.save(c.cpu(), soft_path) + + f0_path = filename + ".f0.npy" + if not os.path.exists(f0_path): + f0 = utils.compute_f0_dio( + wav, sampling_rate=sampling_rate, hop_length=hop_length + ) + np.save(f0_path, f0) + + spec_path = filename.replace(".wav", ".spec.pt") + if not os.path.exists(spec_path): + # Process spectrogram + # The following code can't be replaced by torch.FloatTensor(wav) + # because load_wav_to_torch return a tensor that need to be normalized + + audio, sr = utils.load_wav_to_torch(filename) + if sr != hps.data.sampling_rate: + raise ValueError( + "{} SR doesn't match target {} SR".format( + sr, hps.data.sampling_rate + ) + ) + + audio_norm = audio / hps.data.max_wav_value + audio_norm = audio_norm.unsqueeze(0) + + spec = spectrogram_torch( + audio_norm, + hps.data.filter_length, + hps.data.sampling_rate, + hps.data.hop_length, + hps.data.win_length, + center=False, + ) + spec = torch.squeeze(spec, 0) + torch.save(spec, spec_path) + + +def process_batch(filenames): + print("Loading hubert for content...") + device = "cuda" if torch.cuda.is_available() else "cpu" + hmodel = utils.get_hubert_model().to(device) + print("Loaded hubert.") + for filename in tqdm(filenames): + process_one(filename, hmodel) + + +if __name__ == "__main__": + parser = argparse.ArgumentParser() + parser.add_argument( + "--in_dir", type=str, default="dataset/44k", help="path to input dir" + ) + + args = parser.parse_args() + filenames = glob(f"{args.in_dir}/*/*.wav", recursive=True) # [:10] + shuffle(filenames) + multiprocessing.set_start_method("spawn", force=True) + + num_processes = 1 + chunk_size = int(math.ceil(len(filenames) / num_processes)) + chunks = [ + filenames[i : i + chunk_size] for i in range(0, len(filenames), chunk_size) + ] + print([len(c) for c in chunks]) + processes = [ + multiprocessing.Process(target=process_batch, args=(chunk,)) for chunk in chunks + ] + for p in processes: + p.start() diff --git a/pretrain/nsf_hifigan/put_nsf_hifigan_ckpt_here b/pretrain/nsf_hifigan/put_nsf_hifigan_ckpt_here new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/raw/put_raw_wav_here b/raw/put_raw_wav_here new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/requirements.txt b/requirements.txt new file mode 100644 index 0000000000000000000000000000000000000000..23b57360205cf06942efe4237909628ba4147b41 --- /dev/null +++ b/requirements.txt @@ -0,0 +1,21 @@ +Flask +Flask_Cors +gradio +numpy==1.23.0 +pyworld==0.2.5 +scipy==1.10.0 +SoundFile==0.12.1 +torch==1.13.1 +torchaudio==0.13.1 +torchcrepe +tqdm +scikit-maad +praat-parselmouth +onnx +onnxsim +onnxoptimizer +fairseq==0.12.2 +librosa==0.9.1 +tensorboard +tensorboardX +edge_tts diff --git a/requirements_win.txt b/requirements_win.txt new file mode 100644 index 0000000000000000000000000000000000000000..2c57f89615e79f4ffbfb4b665b6993b209f0da38 --- /dev/null +++ b/requirements_win.txt @@ -0,0 +1,24 @@ +librosa==0.9.1 +fairseq==0.12.2 +Flask==2.1.2 +Flask_Cors==3.0.10 +gradio +numpy +playsound==1.3.0 +PyAudio==0.2.12 +pydub==0.25.1 +pyworld==0.3.0 +requests==2.28.1 +scipy==1.7.3 +sounddevice==0.4.5 +SoundFile==0.10.3.post1 +starlette==0.19.1 +tqdm==4.63.0 +torchcrepe +scikit-maad +praat-parselmouth +onnx +onnxsim +onnxoptimizer +tensorboardX +edge_tts diff --git a/resample.py b/resample.py new file mode 100644 index 0000000000000000000000000000000000000000..b28a86eb779d7b3f163e89fac64ecabe044ad1e2 --- /dev/null +++ b/resample.py @@ -0,0 +1,48 @@ +import os +import argparse +import librosa +import numpy as np +from multiprocessing import Pool, cpu_count +from scipy.io import wavfile +from tqdm import tqdm + + +def process(item): + spkdir, wav_name, args = item + # speaker 's5', 'p280', 'p315' are excluded, + speaker = spkdir.replace("\\", "/").split("/")[-1] + wav_path = os.path.join(args.in_dir, speaker, wav_name) + if os.path.exists(wav_path) and '.wav' in wav_path: + os.makedirs(os.path.join(args.out_dir2, speaker), exist_ok=True) + wav, sr = librosa.load(wav_path, sr=None) + wav, _ = librosa.effects.trim(wav, top_db=20) + peak = np.abs(wav).max() + if peak > 1.0: + wav = 0.98 * wav / peak + wav2 = librosa.resample(wav, orig_sr=sr, target_sr=args.sr2) + wav2 /= max(wav2.max(), -wav2.min()) + save_name = wav_name + save_path2 = os.path.join(args.out_dir2, speaker, save_name) + wavfile.write( + save_path2, + args.sr2, + (wav2 * np.iinfo(np.int16).max).astype(np.int16) + ) + + + +if __name__ == "__main__": + parser = argparse.ArgumentParser() + parser.add_argument("--sr2", type=int, default=44100, help="sampling rate") + parser.add_argument("--in_dir", type=str, default="./dataset_raw", help="path to source dir") + parser.add_argument("--out_dir2", type=str, default="./dataset/44k", help="path to target dir") + args = parser.parse_args() + processs = 30 if cpu_count() > 60 else (cpu_count()-2 if cpu_count() > 4 else 1) + pool = Pool(processes=processs) + + for speaker in os.listdir(args.in_dir): + spk_dir = os.path.join(args.in_dir, speaker) + if os.path.isdir(spk_dir): + print(spk_dir) + for _ in tqdm(pool.imap_unordered(process, [(spk_dir, i, args) for i in os.listdir(spk_dir) if i.endswith("wav")])): + pass diff --git a/train.py b/train.py new file mode 100644 index 0000000000000000000000000000000000000000..9f6e7439961182f54ee880be4b8fa776b44b547f --- /dev/null +++ b/train.py @@ -0,0 +1,315 @@ +import logging +import multiprocessing +import time + +logging.getLogger('matplotlib').setLevel(logging.WARNING) +logging.getLogger('numba').setLevel(logging.WARNING) + +import os +import json +import argparse +import itertools +import math +import torch +from torch import nn, optim +from torch.nn import functional as F +from torch.utils.data import DataLoader +from torch.utils.tensorboard import SummaryWriter +import torch.multiprocessing as mp +import torch.distributed as dist +from torch.nn.parallel import DistributedDataParallel as DDP +from torch.cuda.amp import autocast, GradScaler + +import modules.commons as commons +import utils +from data_utils import TextAudioSpeakerLoader, TextAudioCollate +from models import ( + SynthesizerTrn, + MultiPeriodDiscriminator, +) +from modules.losses import ( + kl_loss, + generator_loss, discriminator_loss, feature_loss +) + +from modules.mel_processing import mel_spectrogram_torch, spec_to_mel_torch + +torch.backends.cudnn.benchmark = True +global_step = 0 +start_time = time.time() + +# os.environ['TORCH_DISTRIBUTED_DEBUG'] = 'INFO' + + +def main(): + """Assume Single Node Multi GPUs Training Only""" + assert torch.cuda.is_available(), "CPU training is not allowed." + hps = utils.get_hparams() + + n_gpus = torch.cuda.device_count() + os.environ['MASTER_ADDR'] = 'localhost' + os.environ['MASTER_PORT'] = hps.train.port + + mp.spawn(run, nprocs=n_gpus, args=(n_gpus, hps,)) + + +def run(rank, n_gpus, hps): + global global_step + if rank == 0: + logger = utils.get_logger(hps.model_dir) + logger.info(hps) + utils.check_git_hash(hps.model_dir) + writer = SummaryWriter(log_dir=hps.model_dir) + writer_eval = SummaryWriter(log_dir=os.path.join(hps.model_dir, "eval")) + + # for pytorch on win, backend use gloo + dist.init_process_group(backend= 'gloo' if os.name == 'nt' else 'nccl', init_method='env://', world_size=n_gpus, rank=rank) + torch.manual_seed(hps.train.seed) + torch.cuda.set_device(rank) + collate_fn = TextAudioCollate() + all_in_mem = hps.train.all_in_mem # If you have enough memory, turn on this option to avoid disk IO and speed up training. + train_dataset = TextAudioSpeakerLoader(hps.data.training_files, hps, all_in_mem=all_in_mem) + num_workers = 5 if multiprocessing.cpu_count() > 4 else multiprocessing.cpu_count() + if all_in_mem: + num_workers = 0 + train_loader = DataLoader(train_dataset, num_workers=num_workers, shuffle=False, pin_memory=True, + batch_size=hps.train.batch_size, collate_fn=collate_fn) + if rank == 0: + eval_dataset = TextAudioSpeakerLoader(hps.data.validation_files, hps, all_in_mem=all_in_mem) + eval_loader = DataLoader(eval_dataset, num_workers=1, shuffle=False, + batch_size=1, pin_memory=False, + drop_last=False, collate_fn=collate_fn) + + net_g = SynthesizerTrn( + hps.data.filter_length // 2 + 1, + hps.train.segment_size // hps.data.hop_length, + **hps.model).cuda(rank) + net_d = MultiPeriodDiscriminator(hps.model.use_spectral_norm).cuda(rank) + optim_g = torch.optim.AdamW( + net_g.parameters(), + hps.train.learning_rate, + betas=hps.train.betas, + eps=hps.train.eps) + optim_d = torch.optim.AdamW( + net_d.parameters(), + hps.train.learning_rate, + betas=hps.train.betas, + eps=hps.train.eps) + net_g = DDP(net_g, device_ids=[rank]) # , find_unused_parameters=True) + net_d = DDP(net_d, device_ids=[rank]) + + skip_optimizer = False + try: + _, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g, + optim_g, skip_optimizer) + _, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "D_*.pth"), net_d, + optim_d, skip_optimizer) + epoch_str = max(epoch_str, 1) + global_step = (epoch_str - 1) * len(train_loader) + except: + print("load old checkpoint failed...") + epoch_str = 1 + global_step = 0 + if skip_optimizer: + epoch_str = 1 + global_step = 0 + + scheduler_g = torch.optim.lr_scheduler.ExponentialLR(optim_g, gamma=hps.train.lr_decay, last_epoch=epoch_str - 2) + scheduler_d = torch.optim.lr_scheduler.ExponentialLR(optim_d, gamma=hps.train.lr_decay, last_epoch=epoch_str - 2) + + scaler = GradScaler(enabled=hps.train.fp16_run) + + for epoch in range(epoch_str, hps.train.epochs + 1): + if rank == 0: + train_and_evaluate(rank, epoch, hps, [net_g, net_d], [optim_g, optim_d], [scheduler_g, scheduler_d], scaler, + [train_loader, eval_loader], logger, [writer, writer_eval]) + else: + train_and_evaluate(rank, epoch, hps, [net_g, net_d], [optim_g, optim_d], [scheduler_g, scheduler_d], scaler, + [train_loader, None], None, None) + scheduler_g.step() + scheduler_d.step() + + +def train_and_evaluate(rank, epoch, hps, nets, optims, schedulers, scaler, loaders, logger, writers): + net_g, net_d = nets + optim_g, optim_d = optims + scheduler_g, scheduler_d = schedulers + train_loader, eval_loader = loaders + if writers is not None: + writer, writer_eval = writers + + # train_loader.batch_sampler.set_epoch(epoch) + global global_step + + net_g.train() + net_d.train() + for batch_idx, items in enumerate(train_loader): + c, f0, spec, y, spk, lengths, uv = items + g = spk.cuda(rank, non_blocking=True) + spec, y = spec.cuda(rank, non_blocking=True), y.cuda(rank, non_blocking=True) + c = c.cuda(rank, non_blocking=True) + f0 = f0.cuda(rank, non_blocking=True) + uv = uv.cuda(rank, non_blocking=True) + lengths = lengths.cuda(rank, non_blocking=True) + mel = spec_to_mel_torch( + spec, + hps.data.filter_length, + hps.data.n_mel_channels, + hps.data.sampling_rate, + hps.data.mel_fmin, + hps.data.mel_fmax) + + with autocast(enabled=hps.train.fp16_run): + y_hat, ids_slice, z_mask, \ + (z, z_p, m_p, logs_p, m_q, logs_q), pred_lf0, norm_lf0, lf0 = net_g(c, f0, uv, spec, g=g, c_lengths=lengths, + spec_lengths=lengths) + + y_mel = commons.slice_segments(mel, ids_slice, hps.train.segment_size // hps.data.hop_length) + y_hat_mel = mel_spectrogram_torch( + y_hat.squeeze(1), + hps.data.filter_length, + hps.data.n_mel_channels, + hps.data.sampling_rate, + hps.data.hop_length, + hps.data.win_length, + hps.data.mel_fmin, + hps.data.mel_fmax + ) + y = commons.slice_segments(y, ids_slice * hps.data.hop_length, hps.train.segment_size) # slice + + # Discriminator + y_d_hat_r, y_d_hat_g, _, _ = net_d(y, y_hat.detach()) + + with autocast(enabled=False): + loss_disc, losses_disc_r, losses_disc_g = discriminator_loss(y_d_hat_r, y_d_hat_g) + loss_disc_all = loss_disc + + optim_d.zero_grad() + scaler.scale(loss_disc_all).backward() + scaler.unscale_(optim_d) + grad_norm_d = commons.clip_grad_value_(net_d.parameters(), None) + scaler.step(optim_d) + + with autocast(enabled=hps.train.fp16_run): + # Generator + y_d_hat_r, y_d_hat_g, fmap_r, fmap_g = net_d(y, y_hat) + with autocast(enabled=False): + loss_mel = F.l1_loss(y_mel, y_hat_mel) * hps.train.c_mel + loss_kl = kl_loss(z_p, logs_q, m_p, logs_p, z_mask) * hps.train.c_kl + loss_fm = feature_loss(fmap_r, fmap_g) + loss_gen, losses_gen = generator_loss(y_d_hat_g) + loss_lf0 = F.mse_loss(pred_lf0, lf0) + loss_gen_all = loss_gen + loss_fm + loss_mel + loss_kl + loss_lf0 + optim_g.zero_grad() + scaler.scale(loss_gen_all).backward() + scaler.unscale_(optim_g) + grad_norm_g = commons.clip_grad_value_(net_g.parameters(), None) + scaler.step(optim_g) + scaler.update() + + if rank == 0: + if global_step % hps.train.log_interval == 0: + lr = optim_g.param_groups[0]['lr'] + losses = [loss_disc, loss_gen, loss_fm, loss_mel, loss_kl] + logger.info('Train Epoch: {} [{:.0f}%]'.format( + epoch, + 100. * batch_idx / len(train_loader))) + logger.info(f"Losses: {[x.item() for x in losses]}, step: {global_step}, lr: {lr}") + + scalar_dict = {"loss/g/total": loss_gen_all, "loss/d/total": loss_disc_all, "learning_rate": lr, + "grad_norm_d": grad_norm_d, "grad_norm_g": grad_norm_g} + scalar_dict.update({"loss/g/fm": loss_fm, "loss/g/mel": loss_mel, "loss/g/kl": loss_kl, + "loss/g/lf0": loss_lf0}) + + # scalar_dict.update({"loss/g/{}".format(i): v for i, v in enumerate(losses_gen)}) + # scalar_dict.update({"loss/d_r/{}".format(i): v for i, v in enumerate(losses_disc_r)}) + # scalar_dict.update({"loss/d_g/{}".format(i): v for i, v in enumerate(losses_disc_g)}) + image_dict = { + "slice/mel_org": utils.plot_spectrogram_to_numpy(y_mel[0].data.cpu().numpy()), + "slice/mel_gen": utils.plot_spectrogram_to_numpy(y_hat_mel[0].data.cpu().numpy()), + "all/mel": utils.plot_spectrogram_to_numpy(mel[0].data.cpu().numpy()), + "all/lf0": utils.plot_data_to_numpy(lf0[0, 0, :].cpu().numpy(), + pred_lf0[0, 0, :].detach().cpu().numpy()), + "all/norm_lf0": utils.plot_data_to_numpy(lf0[0, 0, :].cpu().numpy(), + norm_lf0[0, 0, :].detach().cpu().numpy()) + } + + utils.summarize( + writer=writer, + global_step=global_step, + images=image_dict, + scalars=scalar_dict + ) + + if global_step % hps.train.eval_interval == 0: + evaluate(hps, net_g, eval_loader, writer_eval) + utils.save_checkpoint(net_g, optim_g, hps.train.learning_rate, epoch, + os.path.join(hps.model_dir, "G_{}.pth".format(global_step))) + utils.save_checkpoint(net_d, optim_d, hps.train.learning_rate, epoch, + os.path.join(hps.model_dir, "D_{}.pth".format(global_step))) + keep_ckpts = getattr(hps.train, 'keep_ckpts', 0) + if keep_ckpts > 0: + utils.clean_checkpoints(path_to_models=hps.model_dir, n_ckpts_to_keep=keep_ckpts, sort_by_time=True) + + global_step += 1 + + if rank == 0: + global start_time + now = time.time() + durtaion = format(now - start_time, '.2f') + logger.info(f'====> Epoch: {epoch}, cost {durtaion} s') + start_time = now + + +def evaluate(hps, generator, eval_loader, writer_eval): + generator.eval() + image_dict = {} + audio_dict = {} + with torch.no_grad(): + for batch_idx, items in enumerate(eval_loader): + c, f0, spec, y, spk, _, uv = items + g = spk[:1].cuda(0) + spec, y = spec[:1].cuda(0), y[:1].cuda(0) + c = c[:1].cuda(0) + f0 = f0[:1].cuda(0) + uv= uv[:1].cuda(0) + mel = spec_to_mel_torch( + spec, + hps.data.filter_length, + hps.data.n_mel_channels, + hps.data.sampling_rate, + hps.data.mel_fmin, + hps.data.mel_fmax) + y_hat = generator.module.infer(c, f0, uv, g=g) + + y_hat_mel = mel_spectrogram_torch( + y_hat.squeeze(1).float(), + hps.data.filter_length, + hps.data.n_mel_channels, + hps.data.sampling_rate, + hps.data.hop_length, + hps.data.win_length, + hps.data.mel_fmin, + hps.data.mel_fmax + ) + + audio_dict.update({ + f"gen/audio_{batch_idx}": y_hat[0], + f"gt/audio_{batch_idx}": y[0] + }) + image_dict.update({ + f"gen/mel": utils.plot_spectrogram_to_numpy(y_hat_mel[0].cpu().numpy()), + "gt/mel": utils.plot_spectrogram_to_numpy(mel[0].cpu().numpy()) + }) + utils.summarize( + writer=writer_eval, + global_step=global_step, + images=image_dict, + audios=audio_dict, + audio_sampling_rate=hps.data.sampling_rate + ) + generator.train() + + +if __name__ == "__main__": + main() diff --git a/utils.py b/utils.py new file mode 100644 index 0000000000000000000000000000000000000000..457ae8f4c7b85a4fbd223c3579fc573e5a2abb92 --- /dev/null +++ b/utils.py @@ -0,0 +1,533 @@ +import os +import glob +import re +import sys +import argparse +import logging +import json +import subprocess +import warnings +import random +import functools + +import librosa +import numpy as np +from scipy.io.wavfile import read +import torch +from torch.nn import functional as F +from modules.commons import sequence_mask +from hubert import hubert_model + +MATPLOTLIB_FLAG = False + +logging.basicConfig(stream=sys.stdout, level=logging.DEBUG) +logger = logging + +f0_bin = 256 +f0_max = 1100.0 +f0_min = 50.0 +f0_mel_min = 1127 * np.log(1 + f0_min / 700) +f0_mel_max = 1127 * np.log(1 + f0_max / 700) + + +# def normalize_f0(f0, random_scale=True): +# f0_norm = f0.clone() # create a copy of the input Tensor +# batch_size, _, frame_length = f0_norm.shape +# for i in range(batch_size): +# means = torch.mean(f0_norm[i, 0, :]) +# if random_scale: +# factor = random.uniform(0.8, 1.2) +# else: +# factor = 1 +# f0_norm[i, 0, :] = (f0_norm[i, 0, :] - means) * factor +# return f0_norm +# def normalize_f0(f0, random_scale=True): +# means = torch.mean(f0[:, 0, :], dim=1, keepdim=True) +# if random_scale: +# factor = torch.Tensor(f0.shape[0],1).uniform_(0.8, 1.2).to(f0.device) +# else: +# factor = torch.ones(f0.shape[0], 1, 1).to(f0.device) +# f0_norm = (f0 - means.unsqueeze(-1)) * factor.unsqueeze(-1) +# return f0_norm + +def deprecated(func): + """This is a decorator which can be used to mark functions + as deprecated. It will result in a warning being emitted + when the function is used.""" + @functools.wraps(func) + def new_func(*args, **kwargs): + warnings.simplefilter('always', DeprecationWarning) # turn off filter + warnings.warn("Call to deprecated function {}.".format(func.__name__), + category=DeprecationWarning, + stacklevel=2) + warnings.simplefilter('default', DeprecationWarning) # reset filter + return func(*args, **kwargs) + return new_func + +def normalize_f0(f0, x_mask, uv, random_scale=True): + # calculate means based on x_mask + uv_sum = torch.sum(uv, dim=1, keepdim=True) + uv_sum[uv_sum == 0] = 9999 + means = torch.sum(f0[:, 0, :] * uv, dim=1, keepdim=True) / uv_sum + + if random_scale: + factor = torch.Tensor(f0.shape[0], 1).uniform_(0.8, 1.2).to(f0.device) + else: + factor = torch.ones(f0.shape[0], 1).to(f0.device) + # normalize f0 based on means and factor + f0_norm = (f0 - means.unsqueeze(-1)) * factor.unsqueeze(-1) + if torch.isnan(f0_norm).any(): + exit(0) + return f0_norm * x_mask + +def compute_f0_uv_torchcrepe(wav_numpy, p_len=None, sampling_rate=44100, hop_length=512,device=None): + from modules.crepe import CrepePitchExtractor + x = wav_numpy + if p_len is None: + p_len = x.shape[0]//hop_length + else: + assert abs(p_len-x.shape[0]//hop_length) < 4, "pad length error" + + f0_min = 50 + f0_max = 1100 + F0Creper = CrepePitchExtractor(hop_length=hop_length,f0_min=f0_min,f0_max=f0_max,device=device) + f0,uv = F0Creper(x[None,:].float(),sampling_rate,pad_to=p_len) + return f0,uv + +def plot_data_to_numpy(x, y): + global MATPLOTLIB_FLAG + if not MATPLOTLIB_FLAG: + import matplotlib + matplotlib.use("Agg") + MATPLOTLIB_FLAG = True + mpl_logger = logging.getLogger('matplotlib') + mpl_logger.setLevel(logging.WARNING) + import matplotlib.pylab as plt + import numpy as np + + fig, ax = plt.subplots(figsize=(10, 2)) + plt.plot(x) + plt.plot(y) + plt.tight_layout() + + fig.canvas.draw() + data = np.fromstring(fig.canvas.tostring_rgb(), dtype=np.uint8, sep='') + data = data.reshape(fig.canvas.get_width_height()[::-1] + (3,)) + plt.close() + return data + + + +def interpolate_f0(f0): + ''' + 对F0进行插值处理 + ''' + + data = np.reshape(f0, (f0.size, 1)) + + vuv_vector = np.zeros((data.size, 1), dtype=np.float32) + vuv_vector[data > 0.0] = 1.0 + vuv_vector[data <= 0.0] = 0.0 + + ip_data = data + + frame_number = data.size + last_value = 0.0 + for i in range(frame_number): + if data[i] <= 0.0: + j = i + 1 + for j in range(i + 1, frame_number): + if data[j] > 0.0: + break + if j < frame_number - 1: + if last_value > 0.0: + step = (data[j] - data[i - 1]) / float(j - i) + for k in range(i, j): + ip_data[k] = data[i - 1] + step * (k - i + 1) + else: + for k in range(i, j): + ip_data[k] = data[j] + else: + for k in range(i, frame_number): + ip_data[k] = last_value + else: + ip_data[i] = data[i] #这里可能存在一个没有必要的拷贝 + last_value = data[i] + + return ip_data[:,0], vuv_vector[:,0] + + +def compute_f0_parselmouth(wav_numpy, p_len=None, sampling_rate=44100, hop_length=512): + import parselmouth + x = wav_numpy + if p_len is None: + p_len = x.shape[0]//hop_length + else: + assert abs(p_len-x.shape[0]//hop_length) < 4, "pad length error" + time_step = hop_length / sampling_rate * 1000 + f0_min = 50 + f0_max = 1100 + f0 = parselmouth.Sound(x, sampling_rate).to_pitch_ac( + time_step=time_step / 1000, voicing_threshold=0.6, + pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency'] + + pad_size=(p_len - len(f0) + 1) // 2 + if(pad_size>0 or p_len - len(f0) - pad_size>0): + f0 = np.pad(f0,[[pad_size,p_len - len(f0) - pad_size]], mode='constant') + return f0 + +def resize_f0(x, target_len): + source = np.array(x) + source[source<0.001] = np.nan + target = np.interp(np.arange(0, len(source)*target_len, len(source))/ target_len, np.arange(0, len(source)), source) + res = np.nan_to_num(target) + return res + +def compute_f0_dio(wav_numpy, p_len=None, sampling_rate=44100, hop_length=512): + import pyworld + if p_len is None: + p_len = wav_numpy.shape[0]//hop_length + f0, t = pyworld.dio( + wav_numpy.astype(np.double), + fs=sampling_rate, + f0_ceil=800, + frame_period=1000 * hop_length / sampling_rate, + ) + f0 = pyworld.stonemask(wav_numpy.astype(np.double), f0, t, sampling_rate) + for index, pitch in enumerate(f0): + f0[index] = round(pitch, 1) + return resize_f0(f0, p_len) + +def f0_to_coarse(f0): + is_torch = isinstance(f0, torch.Tensor) + f0_mel = 1127 * (1 + f0 / 700).log() if is_torch else 1127 * np.log(1 + f0 / 700) + f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * (f0_bin - 2) / (f0_mel_max - f0_mel_min) + 1 + + f0_mel[f0_mel <= 1] = 1 + f0_mel[f0_mel > f0_bin - 1] = f0_bin - 1 + f0_coarse = (f0_mel + 0.5).long() if is_torch else np.rint(f0_mel).astype(np.int) + assert f0_coarse.max() <= 255 and f0_coarse.min() >= 1, (f0_coarse.max(), f0_coarse.min()) + return f0_coarse + + +def get_hubert_model(): + vec_path = "hubert/checkpoint_best_legacy_500.pt" + print("load model(s) from {}".format(vec_path)) + from fairseq import checkpoint_utils + models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task( + [vec_path], + suffix="", + ) + model = models[0] + model.eval() + return model + +def get_hubert_content(hmodel, wav_16k_tensor): + feats = wav_16k_tensor + if feats.dim() == 2: # double channels + feats = feats.mean(-1) + assert feats.dim() == 1, feats.dim() + feats = feats.view(1, -1) + padding_mask = torch.BoolTensor(feats.shape).fill_(False) + inputs = { + "source": feats.to(wav_16k_tensor.device), + "padding_mask": padding_mask.to(wav_16k_tensor.device), + "output_layer": 9, # layer 9 + } + with torch.no_grad(): + logits = hmodel.extract_features(**inputs) + feats = hmodel.final_proj(logits[0]) + return feats.transpose(1, 2) + + +def get_content(cmodel, y): + with torch.no_grad(): + c = cmodel.extract_features(y.squeeze(1))[0] + c = c.transpose(1, 2) + return c + + + +def load_checkpoint(checkpoint_path, model, optimizer=None, skip_optimizer=False): + assert os.path.isfile(checkpoint_path) + checkpoint_dict = torch.load(checkpoint_path, map_location='cpu') + iteration = checkpoint_dict['iteration'] + learning_rate = checkpoint_dict['learning_rate'] + if optimizer is not None and not skip_optimizer and checkpoint_dict['optimizer'] is not None: + optimizer.load_state_dict(checkpoint_dict['optimizer']) + saved_state_dict = checkpoint_dict['model'] + if hasattr(model, 'module'): + state_dict = model.module.state_dict() + else: + state_dict = model.state_dict() + new_state_dict = {} + for k, v in state_dict.items(): + try: + # assert "dec" in k or "disc" in k + # print("load", k) + new_state_dict[k] = saved_state_dict[k] + assert saved_state_dict[k].shape == v.shape, (saved_state_dict[k].shape, v.shape) + except: + print("error, %s is not in the checkpoint" % k) + logger.info("%s is not in the checkpoint" % k) + new_state_dict[k] = v + if hasattr(model, 'module'): + model.module.load_state_dict(new_state_dict) + else: + model.load_state_dict(new_state_dict) + print("load ") + logger.info("Loaded checkpoint '{}' (iteration {})".format( + checkpoint_path, iteration)) + return model, optimizer, learning_rate, iteration + + +def save_checkpoint(model, optimizer, learning_rate, iteration, checkpoint_path): + logger.info("Saving model and optimizer state at iteration {} to {}".format( + iteration, checkpoint_path)) + if hasattr(model, 'module'): + state_dict = model.module.state_dict() + else: + state_dict = model.state_dict() + torch.save({'model': state_dict, + 'iteration': iteration, + 'optimizer': optimizer.state_dict(), + 'learning_rate': learning_rate}, checkpoint_path) + +def clean_checkpoints(path_to_models='logs/44k/', n_ckpts_to_keep=2, sort_by_time=True): + """Freeing up space by deleting saved ckpts + + Arguments: + path_to_models -- Path to the model directory + n_ckpts_to_keep -- Number of ckpts to keep, excluding G_0.pth and D_0.pth + sort_by_time -- True -> chronologically delete ckpts + False -> lexicographically delete ckpts + """ + ckpts_files = [f for f in os.listdir(path_to_models) if os.path.isfile(os.path.join(path_to_models, f))] + name_key = (lambda _f: int(re.compile('._(\d+)\.pth').match(_f).group(1))) + time_key = (lambda _f: os.path.getmtime(os.path.join(path_to_models, _f))) + sort_key = time_key if sort_by_time else name_key + x_sorted = lambda _x: sorted([f for f in ckpts_files if f.startswith(_x) and not f.endswith('_0.pth')], key=sort_key) + to_del = [os.path.join(path_to_models, fn) for fn in + (x_sorted('G')[:-n_ckpts_to_keep] + x_sorted('D')[:-n_ckpts_to_keep])] + del_info = lambda fn: logger.info(f".. Free up space by deleting ckpt {fn}") + del_routine = lambda x: [os.remove(x), del_info(x)] + rs = [del_routine(fn) for fn in to_del] + +def summarize(writer, global_step, scalars={}, histograms={}, images={}, audios={}, audio_sampling_rate=22050): + for k, v in scalars.items(): + writer.add_scalar(k, v, global_step) + for k, v in histograms.items(): + writer.add_histogram(k, v, global_step) + for k, v in images.items(): + writer.add_image(k, v, global_step, dataformats='HWC') + for k, v in audios.items(): + writer.add_audio(k, v, global_step, audio_sampling_rate) + + +def latest_checkpoint_path(dir_path, regex="G_*.pth"): + f_list = glob.glob(os.path.join(dir_path, regex)) + f_list.sort(key=lambda f: int("".join(filter(str.isdigit, f)))) + x = f_list[-1] + print(x) + return x + + +def plot_spectrogram_to_numpy(spectrogram): + global MATPLOTLIB_FLAG + if not MATPLOTLIB_FLAG: + import matplotlib + matplotlib.use("Agg") + MATPLOTLIB_FLAG = True + mpl_logger = logging.getLogger('matplotlib') + mpl_logger.setLevel(logging.WARNING) + import matplotlib.pylab as plt + import numpy as np + + fig, ax = plt.subplots(figsize=(10,2)) + im = ax.imshow(spectrogram, aspect="auto", origin="lower", + interpolation='none') + plt.colorbar(im, ax=ax) + plt.xlabel("Frames") + plt.ylabel("Channels") + plt.tight_layout() + + fig.canvas.draw() + data = np.fromstring(fig.canvas.tostring_rgb(), dtype=np.uint8, sep='') + data = data.reshape(fig.canvas.get_width_height()[::-1] + (3,)) + plt.close() + return data + + +def plot_alignment_to_numpy(alignment, info=None): + global MATPLOTLIB_FLAG + if not MATPLOTLIB_FLAG: + import matplotlib + matplotlib.use("Agg") + MATPLOTLIB_FLAG = True + mpl_logger = logging.getLogger('matplotlib') + mpl_logger.setLevel(logging.WARNING) + import matplotlib.pylab as plt + import numpy as np + + fig, ax = plt.subplots(figsize=(6, 4)) + im = ax.imshow(alignment.transpose(), aspect='auto', origin='lower', + interpolation='none') + fig.colorbar(im, ax=ax) + xlabel = 'Decoder timestep' + if info is not None: + xlabel += '\n\n' + info + plt.xlabel(xlabel) + plt.ylabel('Encoder timestep') + plt.tight_layout() + + fig.canvas.draw() + data = np.fromstring(fig.canvas.tostring_rgb(), dtype=np.uint8, sep='') + data = data.reshape(fig.canvas.get_width_height()[::-1] + (3,)) + plt.close() + return data + + +def load_wav_to_torch(full_path): + sampling_rate, data = read(full_path) + return torch.FloatTensor(data.astype(np.float32)), sampling_rate + + +def load_filepaths_and_text(filename, split="|"): + with open(filename, encoding='utf-8') as f: + filepaths_and_text = [line.strip().split(split) for line in f] + return filepaths_and_text + + +def get_hparams(init=True): + parser = argparse.ArgumentParser() + parser.add_argument('-c', '--config', type=str, default="./configs/base.json", + help='JSON file for configuration') + parser.add_argument('-m', '--model', type=str, required=True, + help='Model name') + + args = parser.parse_args() + model_dir = os.path.join("./logs", args.model) + + if not os.path.exists(model_dir): + os.makedirs(model_dir) + + config_path = args.config + config_save_path = os.path.join(model_dir, "config.json") + if init: + with open(config_path, "r") as f: + data = f.read() + with open(config_save_path, "w") as f: + f.write(data) + else: + with open(config_save_path, "r") as f: + data = f.read() + config = json.loads(data) + + hparams = HParams(**config) + hparams.model_dir = model_dir + return hparams + + +def get_hparams_from_dir(model_dir): + config_save_path = os.path.join(model_dir, "config.json") + with open(config_save_path, "r") as f: + data = f.read() + config = json.loads(data) + + hparams =HParams(**config) + hparams.model_dir = model_dir + return hparams + + +def get_hparams_from_file(config_path): + with open(config_path, "r") as f: + data = f.read() + config = json.loads(data) + + hparams =HParams(**config) + return hparams + + +def check_git_hash(model_dir): + source_dir = os.path.dirname(os.path.realpath(__file__)) + if not os.path.exists(os.path.join(source_dir, ".git")): + logger.warn("{} is not a git repository, therefore hash value comparison will be ignored.".format( + source_dir + )) + return + + cur_hash = subprocess.getoutput("git rev-parse HEAD") + + path = os.path.join(model_dir, "githash") + if os.path.exists(path): + saved_hash = open(path).read() + if saved_hash != cur_hash: + logger.warn("git hash values are different. {}(saved) != {}(current)".format( + saved_hash[:8], cur_hash[:8])) + else: + open(path, "w").write(cur_hash) + + +def get_logger(model_dir, filename="train.log"): + global logger + logger = logging.getLogger(os.path.basename(model_dir)) + logger.setLevel(logging.DEBUG) + + formatter = logging.Formatter("%(asctime)s\t%(name)s\t%(levelname)s\t%(message)s") + if not os.path.exists(model_dir): + os.makedirs(model_dir) + h = logging.FileHandler(os.path.join(model_dir, filename)) + h.setLevel(logging.DEBUG) + h.setFormatter(formatter) + logger.addHandler(h) + return logger + + +def repeat_expand_2d(content, target_len): + # content : [h, t] + + src_len = content.shape[-1] + target = torch.zeros([content.shape[0], target_len], dtype=torch.float).to(content.device) + temp = torch.arange(src_len+1) * target_len / src_len + current_pos = 0 + for i in range(target_len): + if i < temp[current_pos+1]: + target[:, i] = content[:, current_pos] + else: + current_pos += 1 + target[:, i] = content[:, current_pos] + + return target + + +class HParams(): + def __init__(self, **kwargs): + for k, v in kwargs.items(): + if type(v) == dict: + v = HParams(**v) + self[k] = v + + def keys(self): + return self.__dict__.keys() + + def items(self): + return self.__dict__.items() + + def values(self): + return self.__dict__.values() + + def __len__(self): + return len(self.__dict__) + + def __getitem__(self, key): + return getattr(self, key) + + def __setitem__(self, key, value): + return setattr(self, key, value) + + def __contains__(self, key): + return key in self.__dict__ + + def __repr__(self): + return self.__dict__.__repr__() + diff --git a/vdecoder/__init__.py b/vdecoder/__init__.py new file mode 100644 index 0000000000000000000000000000000000000000..e69de29bb2d1d6434b8b29ae775ad8c2e48c5391 diff --git a/vdecoder/hifigan/env.py b/vdecoder/hifigan/env.py new file mode 100644 index 0000000000000000000000000000000000000000..2bdbc95d4f7a8bad8fd4f5eef657e2b51d946056 --- /dev/null +++ b/vdecoder/hifigan/env.py @@ -0,0 +1,15 @@ +import os +import shutil + + +class AttrDict(dict): + def __init__(self, *args, **kwargs): + super(AttrDict, self).__init__(*args, **kwargs) + self.__dict__ = self + + +def build_env(config, config_name, path): + t_path = os.path.join(path, config_name) + if config != t_path: + os.makedirs(path, exist_ok=True) + shutil.copyfile(config, os.path.join(path, config_name)) diff --git a/vdecoder/hifigan/models.py b/vdecoder/hifigan/models.py new file mode 100644 index 0000000000000000000000000000000000000000..9747301f350bb269e62601017fe4633ce271b27e --- /dev/null +++ b/vdecoder/hifigan/models.py @@ -0,0 +1,503 @@ +import os +import json +from .env import AttrDict +import numpy as np +import torch +import torch.nn.functional as F +import torch.nn as nn +from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d +from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm +from .utils import init_weights, get_padding + +LRELU_SLOPE = 0.1 + + +def load_model(model_path, device='cuda'): + config_file = os.path.join(os.path.split(model_path)[0], 'config.json') + with open(config_file) as f: + data = f.read() + + global h + json_config = json.loads(data) + h = AttrDict(json_config) + + generator = Generator(h).to(device) + + cp_dict = torch.load(model_path) + generator.load_state_dict(cp_dict['generator']) + generator.eval() + generator.remove_weight_norm() + del cp_dict + return generator, h + + +class ResBlock1(torch.nn.Module): + def __init__(self, h, channels, kernel_size=3, dilation=(1, 3, 5)): + super(ResBlock1, self).__init__() + self.h = h + self.convs1 = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0], + padding=get_padding(kernel_size, dilation[0]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1], + padding=get_padding(kernel_size, dilation[1]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[2], + padding=get_padding(kernel_size, dilation[2]))) + ]) + self.convs1.apply(init_weights) + + self.convs2 = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))) + ]) + self.convs2.apply(init_weights) + + def forward(self, x): + for c1, c2 in zip(self.convs1, self.convs2): + xt = F.leaky_relu(x, LRELU_SLOPE) + xt = c1(xt) + xt = F.leaky_relu(xt, LRELU_SLOPE) + xt = c2(xt) + x = xt + x + return x + + def remove_weight_norm(self): + for l in self.convs1: + remove_weight_norm(l) + for l in self.convs2: + remove_weight_norm(l) + + +class ResBlock2(torch.nn.Module): + def __init__(self, h, channels, kernel_size=3, dilation=(1, 3)): + super(ResBlock2, self).__init__() + self.h = h + self.convs = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0], + padding=get_padding(kernel_size, dilation[0]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1], + padding=get_padding(kernel_size, dilation[1]))) + ]) + self.convs.apply(init_weights) + + def forward(self, x): + for c in self.convs: + xt = F.leaky_relu(x, LRELU_SLOPE) + xt = c(xt) + x = xt + x + return x + + def remove_weight_norm(self): + for l in self.convs: + remove_weight_norm(l) + + +def padDiff(x): + return F.pad(F.pad(x, (0,0,-1,1), 'constant', 0) - x, (0,0,0,-1), 'constant', 0) + +class SineGen(torch.nn.Module): + """ Definition of sine generator + SineGen(samp_rate, harmonic_num = 0, + sine_amp = 0.1, noise_std = 0.003, + voiced_threshold = 0, + flag_for_pulse=False) + samp_rate: sampling rate in Hz + harmonic_num: number of harmonic overtones (default 0) + sine_amp: amplitude of sine-wavefrom (default 0.1) + noise_std: std of Gaussian noise (default 0.003) + voiced_thoreshold: F0 threshold for U/V classification (default 0) + flag_for_pulse: this SinGen is used inside PulseGen (default False) + Note: when flag_for_pulse is True, the first time step of a voiced + segment is always sin(np.pi) or cos(0) + """ + + def __init__(self, samp_rate, harmonic_num=0, + sine_amp=0.1, noise_std=0.003, + voiced_threshold=0, + flag_for_pulse=False): + super(SineGen, self).__init__() + self.sine_amp = sine_amp + self.noise_std = noise_std + self.harmonic_num = harmonic_num + self.dim = self.harmonic_num + 1 + self.sampling_rate = samp_rate + self.voiced_threshold = voiced_threshold + self.flag_for_pulse = flag_for_pulse + + def _f02uv(self, f0): + # generate uv signal + uv = (f0 > self.voiced_threshold).type(torch.float32) + return uv + + def _f02sine(self, f0_values): + """ f0_values: (batchsize, length, dim) + where dim indicates fundamental tone and overtones + """ + # convert to F0 in rad. The interger part n can be ignored + # because 2 * np.pi * n doesn't affect phase + rad_values = (f0_values / self.sampling_rate) % 1 + + # initial phase noise (no noise for fundamental component) + rand_ini = torch.rand(f0_values.shape[0], f0_values.shape[2], \ + device=f0_values.device) + rand_ini[:, 0] = 0 + rad_values[:, 0, :] = rad_values[:, 0, :] + rand_ini + + # instantanouse phase sine[t] = sin(2*pi \sum_i=1 ^{t} rad) + if not self.flag_for_pulse: + # for normal case + + # To prevent torch.cumsum numerical overflow, + # it is necessary to add -1 whenever \sum_k=1^n rad_value_k > 1. + # Buffer tmp_over_one_idx indicates the time step to add -1. + # This will not change F0 of sine because (x-1) * 2*pi = x * 2*pi + tmp_over_one = torch.cumsum(rad_values, 1) % 1 + tmp_over_one_idx = (padDiff(tmp_over_one)) < 0 + cumsum_shift = torch.zeros_like(rad_values) + cumsum_shift[:, 1:, :] = tmp_over_one_idx * -1.0 + + sines = torch.sin(torch.cumsum(rad_values + cumsum_shift, dim=1) + * 2 * np.pi) + else: + # If necessary, make sure that the first time step of every + # voiced segments is sin(pi) or cos(0) + # This is used for pulse-train generation + + # identify the last time step in unvoiced segments + uv = self._f02uv(f0_values) + uv_1 = torch.roll(uv, shifts=-1, dims=1) + uv_1[:, -1, :] = 1 + u_loc = (uv < 1) * (uv_1 > 0) + + # get the instantanouse phase + tmp_cumsum = torch.cumsum(rad_values, dim=1) + # different batch needs to be processed differently + for idx in range(f0_values.shape[0]): + temp_sum = tmp_cumsum[idx, u_loc[idx, :, 0], :] + temp_sum[1:, :] = temp_sum[1:, :] - temp_sum[0:-1, :] + # stores the accumulation of i.phase within + # each voiced segments + tmp_cumsum[idx, :, :] = 0 + tmp_cumsum[idx, u_loc[idx, :, 0], :] = temp_sum + + # rad_values - tmp_cumsum: remove the accumulation of i.phase + # within the previous voiced segment. + i_phase = torch.cumsum(rad_values - tmp_cumsum, dim=1) + + # get the sines + sines = torch.cos(i_phase * 2 * np.pi) + return sines + + def forward(self, f0): + """ sine_tensor, uv = forward(f0) + input F0: tensor(batchsize=1, length, dim=1) + f0 for unvoiced steps should be 0 + output sine_tensor: tensor(batchsize=1, length, dim) + output uv: tensor(batchsize=1, length, 1) + """ + with torch.no_grad(): + f0_buf = torch.zeros(f0.shape[0], f0.shape[1], self.dim, + device=f0.device) + # fundamental component + fn = torch.multiply(f0, torch.FloatTensor([[range(1, self.harmonic_num + 2)]]).to(f0.device)) + + # generate sine waveforms + sine_waves = self._f02sine(fn) * self.sine_amp + + # generate uv signal + # uv = torch.ones(f0.shape) + # uv = uv * (f0 > self.voiced_threshold) + uv = self._f02uv(f0) + + # noise: for unvoiced should be similar to sine_amp + # std = self.sine_amp/3 -> max value ~ self.sine_amp + # . for voiced regions is self.noise_std + noise_amp = uv * self.noise_std + (1 - uv) * self.sine_amp / 3 + noise = noise_amp * torch.randn_like(sine_waves) + + # first: set the unvoiced part to 0 by uv + # then: additive noise + sine_waves = sine_waves * uv + noise + return sine_waves, uv, noise + + +class SourceModuleHnNSF(torch.nn.Module): + """ SourceModule for hn-nsf + SourceModule(sampling_rate, harmonic_num=0, sine_amp=0.1, + add_noise_std=0.003, voiced_threshod=0) + sampling_rate: sampling_rate in Hz + harmonic_num: number of harmonic above F0 (default: 0) + sine_amp: amplitude of sine source signal (default: 0.1) + add_noise_std: std of additive Gaussian noise (default: 0.003) + note that amplitude of noise in unvoiced is decided + by sine_amp + voiced_threshold: threhold to set U/V given F0 (default: 0) + Sine_source, noise_source = SourceModuleHnNSF(F0_sampled) + F0_sampled (batchsize, length, 1) + Sine_source (batchsize, length, 1) + noise_source (batchsize, length 1) + uv (batchsize, length, 1) + """ + + def __init__(self, sampling_rate, harmonic_num=0, sine_amp=0.1, + add_noise_std=0.003, voiced_threshod=0): + super(SourceModuleHnNSF, self).__init__() + + self.sine_amp = sine_amp + self.noise_std = add_noise_std + + # to produce sine waveforms + self.l_sin_gen = SineGen(sampling_rate, harmonic_num, + sine_amp, add_noise_std, voiced_threshod) + + # to merge source harmonics into a single excitation + self.l_linear = torch.nn.Linear(harmonic_num + 1, 1) + self.l_tanh = torch.nn.Tanh() + + def forward(self, x): + """ + Sine_source, noise_source = SourceModuleHnNSF(F0_sampled) + F0_sampled (batchsize, length, 1) + Sine_source (batchsize, length, 1) + noise_source (batchsize, length 1) + """ + # source for harmonic branch + sine_wavs, uv, _ = self.l_sin_gen(x) + sine_merge = self.l_tanh(self.l_linear(sine_wavs)) + + # source for noise branch, in the same shape as uv + noise = torch.randn_like(uv) * self.sine_amp / 3 + return sine_merge, noise, uv + + +class Generator(torch.nn.Module): + def __init__(self, h): + super(Generator, self).__init__() + self.h = h + + self.num_kernels = len(h["resblock_kernel_sizes"]) + self.num_upsamples = len(h["upsample_rates"]) + self.f0_upsamp = torch.nn.Upsample(scale_factor=np.prod(h["upsample_rates"])) + self.m_source = SourceModuleHnNSF( + sampling_rate=h["sampling_rate"], + harmonic_num=8) + self.noise_convs = nn.ModuleList() + self.conv_pre = weight_norm(Conv1d(h["inter_channels"], h["upsample_initial_channel"], 7, 1, padding=3)) + resblock = ResBlock1 if h["resblock"] == '1' else ResBlock2 + self.ups = nn.ModuleList() + for i, (u, k) in enumerate(zip(h["upsample_rates"], h["upsample_kernel_sizes"])): + c_cur = h["upsample_initial_channel"] // (2 ** (i + 1)) + self.ups.append(weight_norm( + ConvTranspose1d(h["upsample_initial_channel"] // (2 ** i), h["upsample_initial_channel"] // (2 ** (i + 1)), + k, u, padding=(k - u) // 2))) + if i + 1 < len(h["upsample_rates"]): # + stride_f0 = np.prod(h["upsample_rates"][i + 1:]) + self.noise_convs.append(Conv1d( + 1, c_cur, kernel_size=stride_f0 * 2, stride=stride_f0, padding=stride_f0 // 2)) + else: + self.noise_convs.append(Conv1d(1, c_cur, kernel_size=1)) + self.resblocks = nn.ModuleList() + for i in range(len(self.ups)): + ch = h["upsample_initial_channel"] // (2 ** (i + 1)) + for j, (k, d) in enumerate(zip(h["resblock_kernel_sizes"], h["resblock_dilation_sizes"])): + self.resblocks.append(resblock(h, ch, k, d)) + + self.conv_post = weight_norm(Conv1d(ch, 1, 7, 1, padding=3)) + self.ups.apply(init_weights) + self.conv_post.apply(init_weights) + self.cond = nn.Conv1d(h['gin_channels'], h['upsample_initial_channel'], 1) + + def forward(self, x, f0, g=None): + # print(1,x.shape,f0.shape,f0[:, None].shape) + f0 = self.f0_upsamp(f0[:, None]).transpose(1, 2) # bs,n,t + # print(2,f0.shape) + har_source, noi_source, uv = self.m_source(f0) + har_source = har_source.transpose(1, 2) + x = self.conv_pre(x) + x = x + self.cond(g) + # print(124,x.shape,har_source.shape) + for i in range(self.num_upsamples): + x = F.leaky_relu(x, LRELU_SLOPE) + # print(3,x.shape) + x = self.ups[i](x) + x_source = self.noise_convs[i](har_source) + # print(4,x_source.shape,har_source.shape,x.shape) + x = x + x_source + xs = None + for j in range(self.num_kernels): + if xs is None: + xs = self.resblocks[i * self.num_kernels + j](x) + else: + xs += self.resblocks[i * self.num_kernels + j](x) + x = xs / self.num_kernels + x = F.leaky_relu(x) + x = self.conv_post(x) + x = torch.tanh(x) + + return x + + def remove_weight_norm(self): + print('Removing weight norm...') + for l in self.ups: + remove_weight_norm(l) + for l in self.resblocks: + l.remove_weight_norm() + remove_weight_norm(self.conv_pre) + remove_weight_norm(self.conv_post) + + +class DiscriminatorP(torch.nn.Module): + def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False): + super(DiscriminatorP, self).__init__() + self.period = period + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(2, 0))), + ]) + self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0))) + + def forward(self, x): + fmap = [] + + # 1d to 2d + b, c, t = x.shape + if t % self.period != 0: # pad first + n_pad = self.period - (t % self.period) + x = F.pad(x, (0, n_pad), "reflect") + t = t + n_pad + x = x.view(b, c, t // self.period, self.period) + + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class MultiPeriodDiscriminator(torch.nn.Module): + def __init__(self, periods=None): + super(MultiPeriodDiscriminator, self).__init__() + self.periods = periods if periods is not None else [2, 3, 5, 7, 11] + self.discriminators = nn.ModuleList() + for period in self.periods: + self.discriminators.append(DiscriminatorP(period)) + + def forward(self, y, y_hat): + y_d_rs = [] + y_d_gs = [] + fmap_rs = [] + fmap_gs = [] + for i, d in enumerate(self.discriminators): + y_d_r, fmap_r = d(y) + y_d_g, fmap_g = d(y_hat) + y_d_rs.append(y_d_r) + fmap_rs.append(fmap_r) + y_d_gs.append(y_d_g) + fmap_gs.append(fmap_g) + + return y_d_rs, y_d_gs, fmap_rs, fmap_gs + + +class DiscriminatorS(torch.nn.Module): + def __init__(self, use_spectral_norm=False): + super(DiscriminatorS, self).__init__() + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv1d(1, 128, 15, 1, padding=7)), + norm_f(Conv1d(128, 128, 41, 2, groups=4, padding=20)), + norm_f(Conv1d(128, 256, 41, 2, groups=16, padding=20)), + norm_f(Conv1d(256, 512, 41, 4, groups=16, padding=20)), + norm_f(Conv1d(512, 1024, 41, 4, groups=16, padding=20)), + norm_f(Conv1d(1024, 1024, 41, 1, groups=16, padding=20)), + norm_f(Conv1d(1024, 1024, 5, 1, padding=2)), + ]) + self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1)) + + def forward(self, x): + fmap = [] + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class MultiScaleDiscriminator(torch.nn.Module): + def __init__(self): + super(MultiScaleDiscriminator, self).__init__() + self.discriminators = nn.ModuleList([ + DiscriminatorS(use_spectral_norm=True), + DiscriminatorS(), + DiscriminatorS(), + ]) + self.meanpools = nn.ModuleList([ + AvgPool1d(4, 2, padding=2), + AvgPool1d(4, 2, padding=2) + ]) + + def forward(self, y, y_hat): + y_d_rs = [] + y_d_gs = [] + fmap_rs = [] + fmap_gs = [] + for i, d in enumerate(self.discriminators): + if i != 0: + y = self.meanpools[i - 1](y) + y_hat = self.meanpools[i - 1](y_hat) + y_d_r, fmap_r = d(y) + y_d_g, fmap_g = d(y_hat) + y_d_rs.append(y_d_r) + fmap_rs.append(fmap_r) + y_d_gs.append(y_d_g) + fmap_gs.append(fmap_g) + + return y_d_rs, y_d_gs, fmap_rs, fmap_gs + + +def feature_loss(fmap_r, fmap_g): + loss = 0 + for dr, dg in zip(fmap_r, fmap_g): + for rl, gl in zip(dr, dg): + loss += torch.mean(torch.abs(rl - gl)) + + return loss * 2 + + +def discriminator_loss(disc_real_outputs, disc_generated_outputs): + loss = 0 + r_losses = [] + g_losses = [] + for dr, dg in zip(disc_real_outputs, disc_generated_outputs): + r_loss = torch.mean((1 - dr) ** 2) + g_loss = torch.mean(dg ** 2) + loss += (r_loss + g_loss) + r_losses.append(r_loss.item()) + g_losses.append(g_loss.item()) + + return loss, r_losses, g_losses + + +def generator_loss(disc_outputs): + loss = 0 + gen_losses = [] + for dg in disc_outputs: + l = torch.mean((1 - dg) ** 2) + gen_losses.append(l) + loss += l + + return loss, gen_losses diff --git a/vdecoder/hifigan/nvSTFT.py b/vdecoder/hifigan/nvSTFT.py new file mode 100644 index 0000000000000000000000000000000000000000..88597d62a505715091f9ba62d38bf0a85a31b95a --- /dev/null +++ b/vdecoder/hifigan/nvSTFT.py @@ -0,0 +1,111 @@ +import math +import os +os.environ["LRU_CACHE_CAPACITY"] = "3" +import random +import torch +import torch.utils.data +import numpy as np +import librosa +from librosa.util import normalize +from librosa.filters import mel as librosa_mel_fn +from scipy.io.wavfile import read +import soundfile as sf + +def load_wav_to_torch(full_path, target_sr=None, return_empty_on_exception=False): + sampling_rate = None + try: + data, sampling_rate = sf.read(full_path, always_2d=True)# than soundfile. + except Exception as ex: + print(f"'{full_path}' failed to load.\nException:") + print(ex) + if return_empty_on_exception: + return [], sampling_rate or target_sr or 32000 + else: + raise Exception(ex) + + if len(data.shape) > 1: + data = data[:, 0] + assert len(data) > 2# check duration of audio file is > 2 samples (because otherwise the slice operation was on the wrong dimension) + + if np.issubdtype(data.dtype, np.integer): # if audio data is type int + max_mag = -np.iinfo(data.dtype).min # maximum magnitude = min possible value of intXX + else: # if audio data is type fp32 + max_mag = max(np.amax(data), -np.amin(data)) + max_mag = (2**31)+1 if max_mag > (2**15) else ((2**15)+1 if max_mag > 1.01 else 1.0) # data should be either 16-bit INT, 32-bit INT or [-1 to 1] float32 + + data = torch.FloatTensor(data.astype(np.float32))/max_mag + + if (torch.isinf(data) | torch.isnan(data)).any() and return_empty_on_exception:# resample will crash with inf/NaN inputs. return_empty_on_exception will return empty arr instead of except + return [], sampling_rate or target_sr or 32000 + if target_sr is not None and sampling_rate != target_sr: + data = torch.from_numpy(librosa.core.resample(data.numpy(), orig_sr=sampling_rate, target_sr=target_sr)) + sampling_rate = target_sr + + return data, sampling_rate + +def dynamic_range_compression(x, C=1, clip_val=1e-5): + return np.log(np.clip(x, a_min=clip_val, a_max=None) * C) + +def dynamic_range_decompression(x, C=1): + return np.exp(x) / C + +def dynamic_range_compression_torch(x, C=1, clip_val=1e-5): + return torch.log(torch.clamp(x, min=clip_val) * C) + +def dynamic_range_decompression_torch(x, C=1): + return torch.exp(x) / C + +class STFT(): + def __init__(self, sr=22050, n_mels=80, n_fft=1024, win_size=1024, hop_length=256, fmin=20, fmax=11025, clip_val=1e-5): + self.target_sr = sr + + self.n_mels = n_mels + self.n_fft = n_fft + self.win_size = win_size + self.hop_length = hop_length + self.fmin = fmin + self.fmax = fmax + self.clip_val = clip_val + self.mel_basis = {} + self.hann_window = {} + + def get_mel(self, y, center=False): + sampling_rate = self.target_sr + n_mels = self.n_mels + n_fft = self.n_fft + win_size = self.win_size + hop_length = self.hop_length + fmin = self.fmin + fmax = self.fmax + clip_val = self.clip_val + + if torch.min(y) < -1.: + print('min value is ', torch.min(y)) + if torch.max(y) > 1.: + print('max value is ', torch.max(y)) + + if fmax not in self.mel_basis: + mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=n_mels, fmin=fmin, fmax=fmax) + self.mel_basis[str(fmax)+'_'+str(y.device)] = torch.from_numpy(mel).float().to(y.device) + self.hann_window[str(y.device)] = torch.hann_window(self.win_size).to(y.device) + + y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_length)/2), int((n_fft-hop_length)/2)), mode='reflect') + y = y.squeeze(1) + + spec = torch.stft(y, n_fft, hop_length=hop_length, win_length=win_size, window=self.hann_window[str(y.device)], + center=center, pad_mode='reflect', normalized=False, onesided=True) + # print(111,spec) + spec = torch.sqrt(spec.pow(2).sum(-1)+(1e-9)) + # print(222,spec) + spec = torch.matmul(self.mel_basis[str(fmax)+'_'+str(y.device)], spec) + # print(333,spec) + spec = dynamic_range_compression_torch(spec, clip_val=clip_val) + # print(444,spec) + return spec + + def __call__(self, audiopath): + audio, sr = load_wav_to_torch(audiopath, target_sr=self.target_sr) + spect = self.get_mel(audio.unsqueeze(0)).squeeze(0) + return spect + +stft = STFT() diff --git a/vdecoder/hifigan/utils.py b/vdecoder/hifigan/utils.py new file mode 100644 index 0000000000000000000000000000000000000000..9c93c996d3cc73c30d71c1fc47056e4230f35c0f --- /dev/null +++ b/vdecoder/hifigan/utils.py @@ -0,0 +1,68 @@ +import glob +import os +import matplotlib +import torch +from torch.nn.utils import weight_norm +# matplotlib.use("Agg") +import matplotlib.pylab as plt + + +def plot_spectrogram(spectrogram): + fig, ax = plt.subplots(figsize=(10, 2)) + im = ax.imshow(spectrogram, aspect="auto", origin="lower", + interpolation='none') + plt.colorbar(im, ax=ax) + + fig.canvas.draw() + plt.close() + + return fig + + +def init_weights(m, mean=0.0, std=0.01): + classname = m.__class__.__name__ + if classname.find("Conv") != -1: + m.weight.data.normal_(mean, std) + + +def apply_weight_norm(m): + classname = m.__class__.__name__ + if classname.find("Conv") != -1: + weight_norm(m) + + +def get_padding(kernel_size, dilation=1): + return int((kernel_size*dilation - dilation)/2) + + +def load_checkpoint(filepath, device): + assert os.path.isfile(filepath) + print("Loading '{}'".format(filepath)) + checkpoint_dict = torch.load(filepath, map_location=device) + print("Complete.") + return checkpoint_dict + + +def save_checkpoint(filepath, obj): + print("Saving checkpoint to {}".format(filepath)) + torch.save(obj, filepath) + print("Complete.") + + +def del_old_checkpoints(cp_dir, prefix, n_models=2): + pattern = os.path.join(cp_dir, prefix + '????????') + cp_list = glob.glob(pattern) # get checkpoint paths + cp_list = sorted(cp_list)# sort by iter + if len(cp_list) > n_models: # if more than n_models models are found + for cp in cp_list[:-n_models]:# delete the oldest models other than lastest n_models + open(cp, 'w').close()# empty file contents + os.unlink(cp)# delete file (move to trash when using Colab) + + +def scan_checkpoint(cp_dir, prefix): + pattern = os.path.join(cp_dir, prefix + '????????') + cp_list = glob.glob(pattern) + if len(cp_list) == 0: + return None + return sorted(cp_list)[-1] + diff --git a/vdecoder/nsf_hifigan/env.py b/vdecoder/nsf_hifigan/env.py new file mode 100644 index 0000000000000000000000000000000000000000..2bdbc95d4f7a8bad8fd4f5eef657e2b51d946056 --- /dev/null +++ b/vdecoder/nsf_hifigan/env.py @@ -0,0 +1,15 @@ +import os +import shutil + + +class AttrDict(dict): + def __init__(self, *args, **kwargs): + super(AttrDict, self).__init__(*args, **kwargs) + self.__dict__ = self + + +def build_env(config, config_name, path): + t_path = os.path.join(path, config_name) + if config != t_path: + os.makedirs(path, exist_ok=True) + shutil.copyfile(config, os.path.join(path, config_name)) diff --git a/vdecoder/nsf_hifigan/models.py b/vdecoder/nsf_hifigan/models.py new file mode 100644 index 0000000000000000000000000000000000000000..6f24f617a76e64bc88b7cff6cc618b59af1c07e3 --- /dev/null +++ b/vdecoder/nsf_hifigan/models.py @@ -0,0 +1,435 @@ +import os +import json +from .env import AttrDict +import numpy as np +import torch +import torch.nn.functional as F +import torch.nn as nn +from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d +from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm +from .utils import init_weights, get_padding + +LRELU_SLOPE = 0.1 + + +def load_model(model_path, device='cuda'): + config_file = os.path.join(os.path.split(model_path)[0], 'config.json') + with open(config_file) as f: + data = f.read() + + json_config = json.loads(data) + h = AttrDict(json_config) + + generator = Generator(h).to(device) + + cp_dict = torch.load(model_path, map_location=device) + generator.load_state_dict(cp_dict['generator']) + generator.eval() + generator.remove_weight_norm() + del cp_dict + return generator, h + + +class ResBlock1(torch.nn.Module): + def __init__(self, h, channels, kernel_size=3, dilation=(1, 3, 5)): + super(ResBlock1, self).__init__() + self.h = h + self.convs1 = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0], + padding=get_padding(kernel_size, dilation[0]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1], + padding=get_padding(kernel_size, dilation[1]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[2], + padding=get_padding(kernel_size, dilation[2]))) + ]) + self.convs1.apply(init_weights) + + self.convs2 = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1, + padding=get_padding(kernel_size, 1))) + ]) + self.convs2.apply(init_weights) + + def forward(self, x): + for c1, c2 in zip(self.convs1, self.convs2): + xt = F.leaky_relu(x, LRELU_SLOPE) + xt = c1(xt) + xt = F.leaky_relu(xt, LRELU_SLOPE) + xt = c2(xt) + x = xt + x + return x + + def remove_weight_norm(self): + for l in self.convs1: + remove_weight_norm(l) + for l in self.convs2: + remove_weight_norm(l) + + +class ResBlock2(torch.nn.Module): + def __init__(self, h, channels, kernel_size=3, dilation=(1, 3)): + super(ResBlock2, self).__init__() + self.h = h + self.convs = nn.ModuleList([ + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0], + padding=get_padding(kernel_size, dilation[0]))), + weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1], + padding=get_padding(kernel_size, dilation[1]))) + ]) + self.convs.apply(init_weights) + + def forward(self, x): + for c in self.convs: + xt = F.leaky_relu(x, LRELU_SLOPE) + xt = c(xt) + x = xt + x + return x + + def remove_weight_norm(self): + for l in self.convs: + remove_weight_norm(l) + + +class SineGen(torch.nn.Module): + """ Definition of sine generator + SineGen(samp_rate, harmonic_num = 0, + sine_amp = 0.1, noise_std = 0.003, + voiced_threshold = 0, + flag_for_pulse=False) + samp_rate: sampling rate in Hz + harmonic_num: number of harmonic overtones (default 0) + sine_amp: amplitude of sine-wavefrom (default 0.1) + noise_std: std of Gaussian noise (default 0.003) + voiced_thoreshold: F0 threshold for U/V classification (default 0) + flag_for_pulse: this SinGen is used inside PulseGen (default False) + Note: when flag_for_pulse is True, the first time step of a voiced + segment is always sin(np.pi) or cos(0) + """ + + def __init__(self, samp_rate, harmonic_num=0, + sine_amp=0.1, noise_std=0.003, + voiced_threshold=0): + super(SineGen, self).__init__() + self.sine_amp = sine_amp + self.noise_std = noise_std + self.harmonic_num = harmonic_num + self.dim = self.harmonic_num + 1 + self.sampling_rate = samp_rate + self.voiced_threshold = voiced_threshold + + def _f02uv(self, f0): + # generate uv signal + uv = torch.ones_like(f0) + uv = uv * (f0 > self.voiced_threshold) + return uv + + @torch.no_grad() + def forward(self, f0, upp): + """ sine_tensor, uv = forward(f0) + input F0: tensor(batchsize=1, length, dim=1) + f0 for unvoiced steps should be 0 + output sine_tensor: tensor(batchsize=1, length, dim) + output uv: tensor(batchsize=1, length, 1) + """ + f0 = f0.unsqueeze(-1) + fn = torch.multiply(f0, torch.arange(1, self.dim + 1, device=f0.device).reshape((1, 1, -1))) + rad_values = (fn / self.sampling_rate) % 1 ###%1意味着n_har的乘积无法后处理优化 + rand_ini = torch.rand(fn.shape[0], fn.shape[2], device=fn.device) + rand_ini[:, 0] = 0 + rad_values[:, 0, :] = rad_values[:, 0, :] + rand_ini + is_half = rad_values.dtype is not torch.float32 + tmp_over_one = torch.cumsum(rad_values.double(), 1) # % 1 #####%1意味着后面的cumsum无法再优化 + if is_half: + tmp_over_one = tmp_over_one.half() + else: + tmp_over_one = tmp_over_one.float() + tmp_over_one *= upp + tmp_over_one = F.interpolate( + tmp_over_one.transpose(2, 1), scale_factor=upp, + mode='linear', align_corners=True + ).transpose(2, 1) + rad_values = F.interpolate(rad_values.transpose(2, 1), scale_factor=upp, mode='nearest').transpose(2, 1) + tmp_over_one %= 1 + tmp_over_one_idx = (tmp_over_one[:, 1:, :] - tmp_over_one[:, :-1, :]) < 0 + cumsum_shift = torch.zeros_like(rad_values) + cumsum_shift[:, 1:, :] = tmp_over_one_idx * -1.0 + rad_values = rad_values.double() + cumsum_shift = cumsum_shift.double() + sine_waves = torch.sin(torch.cumsum(rad_values + cumsum_shift, dim=1) * 2 * np.pi) + if is_half: + sine_waves = sine_waves.half() + else: + sine_waves = sine_waves.float() + sine_waves = sine_waves * self.sine_amp + uv = self._f02uv(f0) + uv = F.interpolate(uv.transpose(2, 1), scale_factor=upp, mode='nearest').transpose(2, 1) + noise_amp = uv * self.noise_std + (1 - uv) * self.sine_amp / 3 + noise = noise_amp * torch.randn_like(sine_waves) + sine_waves = sine_waves * uv + noise + return sine_waves, uv, noise + + +class SourceModuleHnNSF(torch.nn.Module): + """ SourceModule for hn-nsf + SourceModule(sampling_rate, harmonic_num=0, sine_amp=0.1, + add_noise_std=0.003, voiced_threshod=0) + sampling_rate: sampling_rate in Hz + harmonic_num: number of harmonic above F0 (default: 0) + sine_amp: amplitude of sine source signal (default: 0.1) + add_noise_std: std of additive Gaussian noise (default: 0.003) + note that amplitude of noise in unvoiced is decided + by sine_amp + voiced_threshold: threhold to set U/V given F0 (default: 0) + Sine_source, noise_source = SourceModuleHnNSF(F0_sampled) + F0_sampled (batchsize, length, 1) + Sine_source (batchsize, length, 1) + noise_source (batchsize, length 1) + uv (batchsize, length, 1) + """ + + def __init__(self, sampling_rate, harmonic_num=0, sine_amp=0.1, + add_noise_std=0.003, voiced_threshod=0): + super(SourceModuleHnNSF, self).__init__() + + self.sine_amp = sine_amp + self.noise_std = add_noise_std + + # to produce sine waveforms + self.l_sin_gen = SineGen(sampling_rate, harmonic_num, + sine_amp, add_noise_std, voiced_threshod) + + # to merge source harmonics into a single excitation + self.l_linear = torch.nn.Linear(harmonic_num + 1, 1) + self.l_tanh = torch.nn.Tanh() + + def forward(self, x, upp): + sine_wavs, uv, _ = self.l_sin_gen(x, upp) + sine_merge = self.l_tanh(self.l_linear(sine_wavs)) + return sine_merge + + +class Generator(torch.nn.Module): + def __init__(self, h): + super(Generator, self).__init__() + self.h = h + self.num_kernels = len(h.resblock_kernel_sizes) + self.num_upsamples = len(h.upsample_rates) + self.m_source = SourceModuleHnNSF( + sampling_rate=h.sampling_rate, + harmonic_num=8 + ) + self.noise_convs = nn.ModuleList() + self.conv_pre = weight_norm(Conv1d(h.num_mels, h.upsample_initial_channel, 7, 1, padding=3)) + resblock = ResBlock1 if h.resblock == '1' else ResBlock2 + + self.ups = nn.ModuleList() + for i, (u, k) in enumerate(zip(h.upsample_rates, h.upsample_kernel_sizes)): + c_cur = h.upsample_initial_channel // (2 ** (i + 1)) + self.ups.append(weight_norm( + ConvTranspose1d(h.upsample_initial_channel // (2 ** i), h.upsample_initial_channel // (2 ** (i + 1)), + k, u, padding=(k - u) // 2))) + if i + 1 < len(h.upsample_rates): # + stride_f0 = int(np.prod(h.upsample_rates[i + 1:])) + self.noise_convs.append(Conv1d( + 1, c_cur, kernel_size=stride_f0 * 2, stride=stride_f0, padding=stride_f0 // 2)) + else: + self.noise_convs.append(Conv1d(1, c_cur, kernel_size=1)) + self.resblocks = nn.ModuleList() + ch = h.upsample_initial_channel + for i in range(len(self.ups)): + ch //= 2 + for j, (k, d) in enumerate(zip(h.resblock_kernel_sizes, h.resblock_dilation_sizes)): + self.resblocks.append(resblock(h, ch, k, d)) + + self.conv_post = weight_norm(Conv1d(ch, 1, 7, 1, padding=3)) + self.ups.apply(init_weights) + self.conv_post.apply(init_weights) + self.upp = int(np.prod(h.upsample_rates)) + + def forward(self, x, f0): + har_source = self.m_source(f0, self.upp).transpose(1, 2) + x = self.conv_pre(x) + for i in range(self.num_upsamples): + x = F.leaky_relu(x, LRELU_SLOPE) + x = self.ups[i](x) + x_source = self.noise_convs[i](har_source) + x = x + x_source + xs = None + for j in range(self.num_kernels): + if xs is None: + xs = self.resblocks[i * self.num_kernels + j](x) + else: + xs += self.resblocks[i * self.num_kernels + j](x) + x = xs / self.num_kernels + x = F.leaky_relu(x) + x = self.conv_post(x) + x = torch.tanh(x) + + return x + + def remove_weight_norm(self): + print('Removing weight norm...') + for l in self.ups: + remove_weight_norm(l) + for l in self.resblocks: + l.remove_weight_norm() + remove_weight_norm(self.conv_pre) + remove_weight_norm(self.conv_post) + + +class DiscriminatorP(torch.nn.Module): + def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False): + super(DiscriminatorP, self).__init__() + self.period = period + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))), + norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(2, 0))), + ]) + self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0))) + + def forward(self, x): + fmap = [] + + # 1d to 2d + b, c, t = x.shape + if t % self.period != 0: # pad first + n_pad = self.period - (t % self.period) + x = F.pad(x, (0, n_pad), "reflect") + t = t + n_pad + x = x.view(b, c, t // self.period, self.period) + + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class MultiPeriodDiscriminator(torch.nn.Module): + def __init__(self, periods=None): + super(MultiPeriodDiscriminator, self).__init__() + self.periods = periods if periods is not None else [2, 3, 5, 7, 11] + self.discriminators = nn.ModuleList() + for period in self.periods: + self.discriminators.append(DiscriminatorP(period)) + + def forward(self, y, y_hat): + y_d_rs = [] + y_d_gs = [] + fmap_rs = [] + fmap_gs = [] + for i, d in enumerate(self.discriminators): + y_d_r, fmap_r = d(y) + y_d_g, fmap_g = d(y_hat) + y_d_rs.append(y_d_r) + fmap_rs.append(fmap_r) + y_d_gs.append(y_d_g) + fmap_gs.append(fmap_g) + + return y_d_rs, y_d_gs, fmap_rs, fmap_gs + + +class DiscriminatorS(torch.nn.Module): + def __init__(self, use_spectral_norm=False): + super(DiscriminatorS, self).__init__() + norm_f = weight_norm if use_spectral_norm == False else spectral_norm + self.convs = nn.ModuleList([ + norm_f(Conv1d(1, 128, 15, 1, padding=7)), + norm_f(Conv1d(128, 128, 41, 2, groups=4, padding=20)), + norm_f(Conv1d(128, 256, 41, 2, groups=16, padding=20)), + norm_f(Conv1d(256, 512, 41, 4, groups=16, padding=20)), + norm_f(Conv1d(512, 1024, 41, 4, groups=16, padding=20)), + norm_f(Conv1d(1024, 1024, 41, 1, groups=16, padding=20)), + norm_f(Conv1d(1024, 1024, 5, 1, padding=2)), + ]) + self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1)) + + def forward(self, x): + fmap = [] + for l in self.convs: + x = l(x) + x = F.leaky_relu(x, LRELU_SLOPE) + fmap.append(x) + x = self.conv_post(x) + fmap.append(x) + x = torch.flatten(x, 1, -1) + + return x, fmap + + +class MultiScaleDiscriminator(torch.nn.Module): + def __init__(self): + super(MultiScaleDiscriminator, self).__init__() + self.discriminators = nn.ModuleList([ + DiscriminatorS(use_spectral_norm=True), + DiscriminatorS(), + DiscriminatorS(), + ]) + self.meanpools = nn.ModuleList([ + AvgPool1d(4, 2, padding=2), + AvgPool1d(4, 2, padding=2) + ]) + + def forward(self, y, y_hat): + y_d_rs = [] + y_d_gs = [] + fmap_rs = [] + fmap_gs = [] + for i, d in enumerate(self.discriminators): + if i != 0: + y = self.meanpools[i - 1](y) + y_hat = self.meanpools[i - 1](y_hat) + y_d_r, fmap_r = d(y) + y_d_g, fmap_g = d(y_hat) + y_d_rs.append(y_d_r) + fmap_rs.append(fmap_r) + y_d_gs.append(y_d_g) + fmap_gs.append(fmap_g) + + return y_d_rs, y_d_gs, fmap_rs, fmap_gs + + +def feature_loss(fmap_r, fmap_g): + loss = 0 + for dr, dg in zip(fmap_r, fmap_g): + for rl, gl in zip(dr, dg): + loss += torch.mean(torch.abs(rl - gl)) + + return loss * 2 + + +def discriminator_loss(disc_real_outputs, disc_generated_outputs): + loss = 0 + r_losses = [] + g_losses = [] + for dr, dg in zip(disc_real_outputs, disc_generated_outputs): + r_loss = torch.mean((1 - dr) ** 2) + g_loss = torch.mean(dg ** 2) + loss += (r_loss + g_loss) + r_losses.append(r_loss.item()) + g_losses.append(g_loss.item()) + + return loss, r_losses, g_losses + + +def generator_loss(disc_outputs): + loss = 0 + gen_losses = [] + for dg in disc_outputs: + l = torch.mean((1 - dg) ** 2) + gen_losses.append(l) + loss += l + + return loss, gen_losses diff --git a/vdecoder/nsf_hifigan/nvSTFT.py b/vdecoder/nsf_hifigan/nvSTFT.py new file mode 100644 index 0000000000000000000000000000000000000000..62bd5a008f81929054f036c81955d5d73377f772 --- /dev/null +++ b/vdecoder/nsf_hifigan/nvSTFT.py @@ -0,0 +1,134 @@ +import math +import os +os.environ["LRU_CACHE_CAPACITY"] = "3" +import random +import torch +import torch.utils.data +import numpy as np +import librosa +from librosa.util import normalize +from librosa.filters import mel as librosa_mel_fn +from scipy.io.wavfile import read +import soundfile as sf +import torch.nn.functional as F + +def load_wav_to_torch(full_path, target_sr=None, return_empty_on_exception=False): + sampling_rate = None + try: + data, sampling_rate = sf.read(full_path, always_2d=True)# than soundfile. + except Exception as ex: + print(f"'{full_path}' failed to load.\nException:") + print(ex) + if return_empty_on_exception: + return [], sampling_rate or target_sr or 48000 + else: + raise Exception(ex) + + if len(data.shape) > 1: + data = data[:, 0] + assert len(data) > 2# check duration of audio file is > 2 samples (because otherwise the slice operation was on the wrong dimension) + + if np.issubdtype(data.dtype, np.integer): # if audio data is type int + max_mag = -np.iinfo(data.dtype).min # maximum magnitude = min possible value of intXX + else: # if audio data is type fp32 + max_mag = max(np.amax(data), -np.amin(data)) + max_mag = (2**31)+1 if max_mag > (2**15) else ((2**15)+1 if max_mag > 1.01 else 1.0) # data should be either 16-bit INT, 32-bit INT or [-1 to 1] float32 + + data = torch.FloatTensor(data.astype(np.float32))/max_mag + + if (torch.isinf(data) | torch.isnan(data)).any() and return_empty_on_exception:# resample will crash with inf/NaN inputs. return_empty_on_exception will return empty arr instead of except + return [], sampling_rate or target_sr or 48000 + if target_sr is not None and sampling_rate != target_sr: + data = torch.from_numpy(librosa.core.resample(data.numpy(), orig_sr=sampling_rate, target_sr=target_sr)) + sampling_rate = target_sr + + return data, sampling_rate + +def dynamic_range_compression(x, C=1, clip_val=1e-5): + return np.log(np.clip(x, a_min=clip_val, a_max=None) * C) + +def dynamic_range_decompression(x, C=1): + return np.exp(x) / C + +def dynamic_range_compression_torch(x, C=1, clip_val=1e-5): + return torch.log(torch.clamp(x, min=clip_val) * C) + +def dynamic_range_decompression_torch(x, C=1): + return torch.exp(x) / C + +class STFT(): + def __init__(self, sr=22050, n_mels=80, n_fft=1024, win_size=1024, hop_length=256, fmin=20, fmax=11025, clip_val=1e-5): + self.target_sr = sr + + self.n_mels = n_mels + self.n_fft = n_fft + self.win_size = win_size + self.hop_length = hop_length + self.fmin = fmin + self.fmax = fmax + self.clip_val = clip_val + self.mel_basis = {} + self.hann_window = {} + + def get_mel(self, y, keyshift=0, speed=1, center=False): + sampling_rate = self.target_sr + n_mels = self.n_mels + n_fft = self.n_fft + win_size = self.win_size + hop_length = self.hop_length + fmin = self.fmin + fmax = self.fmax + clip_val = self.clip_val + + factor = 2 ** (keyshift / 12) + n_fft_new = int(np.round(n_fft * factor)) + win_size_new = int(np.round(win_size * factor)) + hop_length_new = int(np.round(hop_length * speed)) + + if torch.min(y) < -1.: + print('min value is ', torch.min(y)) + if torch.max(y) > 1.: + print('max value is ', torch.max(y)) + + mel_basis_key = str(fmax)+'_'+str(y.device) + if mel_basis_key not in self.mel_basis: + mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=n_mels, fmin=fmin, fmax=fmax) + self.mel_basis[mel_basis_key] = torch.from_numpy(mel).float().to(y.device) + + keyshift_key = str(keyshift)+'_'+str(y.device) + if keyshift_key not in self.hann_window: + self.hann_window[keyshift_key] = torch.hann_window(win_size_new).to(y.device) + + pad_left = (win_size_new - hop_length_new) //2 + pad_right = max((win_size_new- hop_length_new + 1) //2, win_size_new - y.size(-1) - pad_left) + if pad_right < y.size(-1): + mode = 'reflect' + else: + mode = 'constant' + y = torch.nn.functional.pad(y.unsqueeze(1), (pad_left, pad_right), mode = mode) + y = y.squeeze(1) + + spec = torch.stft(y, n_fft_new, hop_length=hop_length_new, win_length=win_size_new, window=self.hann_window[keyshift_key], + center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=False) + # print(111,spec) + spec = torch.sqrt(spec.pow(2).sum(-1)+(1e-9)) + if keyshift != 0: + size = n_fft // 2 + 1 + resize = spec.size(1) + if resize < size: + spec = F.pad(spec, (0, 0, 0, size-resize)) + spec = spec[:, :size, :] * win_size / win_size_new + + # print(222,spec) + spec = torch.matmul(self.mel_basis[mel_basis_key], spec) + # print(333,spec) + spec = dynamic_range_compression_torch(spec, clip_val=clip_val) + # print(444,spec) + return spec + + def __call__(self, audiopath): + audio, sr = load_wav_to_torch(audiopath, target_sr=self.target_sr) + spect = self.get_mel(audio.unsqueeze(0)).squeeze(0) + return spect + +stft = STFT() diff --git a/vdecoder/nsf_hifigan/utils.py b/vdecoder/nsf_hifigan/utils.py new file mode 100644 index 0000000000000000000000000000000000000000..84bff024f4d2e2de194b2a88ee7bbe5f0d33f67c --- /dev/null +++ b/vdecoder/nsf_hifigan/utils.py @@ -0,0 +1,68 @@ +import glob +import os +import matplotlib +import torch +from torch.nn.utils import weight_norm +matplotlib.use("Agg") +import matplotlib.pylab as plt + + +def plot_spectrogram(spectrogram): + fig, ax = plt.subplots(figsize=(10, 2)) + im = ax.imshow(spectrogram, aspect="auto", origin="lower", + interpolation='none') + plt.colorbar(im, ax=ax) + + fig.canvas.draw() + plt.close() + + return fig + + +def init_weights(m, mean=0.0, std=0.01): + classname = m.__class__.__name__ + if classname.find("Conv") != -1: + m.weight.data.normal_(mean, std) + + +def apply_weight_norm(m): + classname = m.__class__.__name__ + if classname.find("Conv") != -1: + weight_norm(m) + + +def get_padding(kernel_size, dilation=1): + return int((kernel_size*dilation - dilation)/2) + + +def load_checkpoint(filepath, device): + assert os.path.isfile(filepath) + print("Loading '{}'".format(filepath)) + checkpoint_dict = torch.load(filepath, map_location=device) + print("Complete.") + return checkpoint_dict + + +def save_checkpoint(filepath, obj): + print("Saving checkpoint to {}".format(filepath)) + torch.save(obj, filepath) + print("Complete.") + + +def del_old_checkpoints(cp_dir, prefix, n_models=2): + pattern = os.path.join(cp_dir, prefix + '????????') + cp_list = glob.glob(pattern) # get checkpoint paths + cp_list = sorted(cp_list)# sort by iter + if len(cp_list) > n_models: # if more than n_models models are found + for cp in cp_list[:-n_models]:# delete the oldest models other than lastest n_models + open(cp, 'w').close()# empty file contents + os.unlink(cp)# delete file (move to trash when using Colab) + + +def scan_checkpoint(cp_dir, prefix): + pattern = os.path.join(cp_dir, prefix + '????????') + cp_list = glob.glob(pattern) + if len(cp_list) == 0: + return None + return sorted(cp_list)[-1] + diff --git a/wav_upload.py b/wav_upload.py new file mode 100644 index 0000000000000000000000000000000000000000..cac679de78634e638e9a998615406b1c36374fb5 --- /dev/null +++ b/wav_upload.py @@ -0,0 +1,23 @@ +from google.colab import files +import shutil +import os +import argparse +if __name__ == "__main__": + parser = argparse.ArgumentParser() + parser.add_argument("--type", type=str, required=True, help="type of file to upload") + args = parser.parse_args() + file_type = args.type + + basepath = os.getcwd() + uploaded = files.upload() # 上传文件 + assert(file_type in ['zip', 'audio']) + if file_type == "zip": + upload_path = "./upload/" + for filename in uploaded.keys(): + #将上传的文件移动到指定的位置上 + shutil.move(os.path.join(basepath, filename), os.path.join(upload_path, "userzip.zip")) + elif file_type == "audio": + upload_path = "./raw/" + for filename in uploaded.keys(): + #将上传的文件移动到指定的位置上 + shutil.move(os.path.join(basepath, filename), os.path.join(upload_path, filename)) \ No newline at end of file diff --git a/webUI.py b/webUI.py new file mode 100644 index 0000000000000000000000000000000000000000..c0467bae07a7272a4c6b6d647d4c642a1f27d967 --- /dev/null +++ b/webUI.py @@ -0,0 +1,186 @@ +import io +import os + +# os.system("wget -P cvec/ https://huggingface.co/spaces/innnky/nanami/resolve/main/checkpoint_best_legacy_500.pt") +import gradio as gr +import gradio.processing_utils as gr_pu +import librosa +import numpy as np +import soundfile +from inference.infer_tool import Svc +import logging +import traceback + +import subprocess +import edge_tts +import asyncio +from scipy.io import wavfile +import librosa +import torch +import time + +logging.getLogger('numba').setLevel(logging.WARNING) +logging.getLogger('markdown_it').setLevel(logging.WARNING) +logging.getLogger('urllib3').setLevel(logging.WARNING) +logging.getLogger('matplotlib').setLevel(logging.WARNING) +logging.getLogger('multipart').setLevel(logging.WARNING) + +model = None +spk = None +debug=False + +cuda = [] +if torch.cuda.is_available(): + for i in range(torch.cuda.device_count()): + cuda.append("cuda:{}".format(i)) + +def vc_fn(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key): + global model + try: + if input_audio is None: + return "You need to upload an audio", None + if model is None: + return "You need to upload an model", None + sampling_rate, audio = input_audio + # print(audio.shape,sampling_rate) + audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32) + if len(audio.shape) > 1: + audio = librosa.to_mono(audio.transpose(1, 0)) + temp_path = "temp.wav" + soundfile.write(temp_path, audio, sampling_rate, format="wav") + _audio = model.slice_inference(temp_path, sid, vc_transform, slice_db, cluster_ratio, auto_f0, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key) + model.clear_empty() + os.remove(temp_path) + #构建保存文件的路径，并保存到results文件夹内 + try: + timestamp = str(int(time.time())) + output_file = os.path.join("./results", sid + "_" + timestamp + ".wav") + soundfile.write(output_file, _audio, model.target_sample, format="wav") + return "Success", (model.target_sample, _audio) + except Exception as e: + if debug:traceback.print_exc() + return "自动保存失败，请手动保存，音乐输出见下", (model.target_sample, _audio) + except Exception as e: + if debug:traceback.print_exc() + return "异常信息:"+str(e)+"\n请排障后重试",None + +def tts_func(_text,_rate): + #使用edge-tts把文字转成音频 + # voice = "zh-CN-XiaoyiNeural"#女性，较高音 + # voice = "zh-CN-YunxiNeural"#男性 + voice = "zh-CN-YunxiNeural"#男性 + output_file = _text[0:10]+".wav" + # communicate = edge_tts.Communicate(_text, voice) + # await communicate.save(output_file) + if _rate>=0: + ratestr="+{:.0%}".format(_rate) + elif _rate<0: + ratestr="{:.0%}".format(_rate)#减号自带 + + p=subprocess.Popen(["edge-tts", + "--text",_text, + "--write-media",output_file, + "--voice",voice, + "--rate="+ratestr] + ,shell=True, + stdout=subprocess.PIPE, + stdin=subprocess.PIPE) + p.wait() + return output_file + +def vc_fn2(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,text2tts,tts_rate,F0_mean_pooling,enhancer_adaptive_key): + #使用edge-tts把文字转成音频 + output_file=tts_func(text2tts,tts_rate) + + #调整采样率 + sr2=44100 + wav, sr = librosa.load(output_file) + wav2 = librosa.resample(wav, orig_sr=sr, target_sr=sr2) + save_path2= text2tts[0:10]+"_44k"+".wav" + wavfile.write(save_path2,sr2, + (wav2 * np.iinfo(np.int16).max).astype(np.int16) + ) + + #读取音频 + sample_rate, data=gr_pu.audio_from_file(save_path2) + vc_input=(sample_rate, data) + + a,b=vc_fn(sid, vc_input, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key) + os.remove(output_file) + os.remove(save_path2) + return a,b + +app = gr.Blocks() +with app: + with gr.Tabs(): + with gr.TabItem("Sovits4.0"): + gr.Markdown(value=""" + Sovits4.0 WebUI + """) + + gr.Markdown(value=""" + 下面是模型文件选择： + """) + model_path = gr.File(label="模型文件") + gr.Markdown(value=""" + 下面是配置文件选择： + """) + config_path = gr.File(label="配置文件") + gr.Markdown(value=""" + 下面是聚类模型文件选择，没有可以不填： + """) + cluster_model_path = gr.File(label="聚类模型文件") + device = gr.Dropdown(label="推理设备，默认为自动选择cpu和gpu",choices=["Auto",*cuda,"cpu"],value="Auto") + enhance = gr.Checkbox(label="是否使用NSF_HIFIGAN增强,该选项对部分训练集少的模型有一定的音质增强效果，但是对训练好的模型有反面效果，默认关闭", value=False) + gr.Markdown(value=""" + 全部上传完毕后(全部文件模块显示download),点击模型解析进行解析： + """) + model_analysis_button = gr.Button(value="模型解析") + model_unload_button = gr.Button(value="模型卸载") + sid = gr.Dropdown(label="音色（说话人）") + sid_output = gr.Textbox(label="Output Message") + + text2tts=gr.Textbox(label="在此输入要转译的文字。注意，使用该功能建议打开F0预测，不然会很怪") + tts_rate = gr.Number(label="tts语速", value=0) + + vc_input3 = gr.Audio(label="上传音频") + vc_transform = gr.Number(label="变调（整数，可以正负，半音数量，升高八度就是12）", value=0) + cluster_ratio = gr.Number(label="聚类模型混合比例，0-1之间，默认为0不启用聚类，能提升音色相似度，但会导致咬字下降（如果使用建议0.5左右）", value=0) + auto_f0 = gr.Checkbox(label="自动f0预测，配合聚类模型f0预测效果更好,会导致变调功能失效（仅限转换语音，歌声不要勾选此项会究极跑调）", value=False) + F0_mean_pooling = gr.Checkbox(label="是否对F0使用均值滤波器(池化)，对部分哑音有改善。注意，启动该选项会导致推理速度下降，默认关闭", value=False) + slice_db = gr.Number(label="切片阈值", value=-40) + noise_scale = gr.Number(label="noise_scale 建议不要动，会影响音质，玄学参数", value=0.4) + cl_num = gr.Number(label="音频自动切片，0为不切片，单位为秒/s", value=0) + pad_seconds = gr.Number(label="推理音频pad秒数，由于未知原因开头结尾会有异响，pad一小段静音段后就不会出现", value=0.5) + lg_num = gr.Number(label="两端音频切片的交叉淡入长度，如果自动切片后出现人声不连贯可调整该数值，如果连贯建议采用默认值0，注意，该设置会影响推理速度，单位为秒/s", value=0) + lgr_num = gr.Number(label="自动音频切片后，需要舍弃每段切片的头尾。该参数设置交叉长度保留的比例，范围0-1,左开右闭", value=0.75,interactive=True) + enhancer_adaptive_key = gr.Number(label="使增强器适应更高的音域(单位为半音数)|默认为0", value=0,interactive=True) + vc_submit = gr.Button("音频直接转换", variant="primary") + vc_submit2 = gr.Button("文字转音频+转换", variant="primary") + vc_output1 = gr.Textbox(label="Output Message") + vc_output2 = gr.Audio(label="Output Audio") + def modelAnalysis(model_path,config_path,cluster_model_path,device,enhance): + global model + try: + model = Svc(model_path.name, config_path.name,device=device if device!="Auto" else None,cluster_model_path= cluster_model_path.name if cluster_model_path!=None else "",nsf_hifigan_enhance=enhance) + spks = list(model.spk2id.keys()) + device_name = torch.cuda.get_device_properties(model.dev).name if "cuda" in str(model.dev) else str(model.dev) + return sid.update(choices = spks,value=spks[0]),"ok,模型被加载到了设备{}之上".format(device_name) + except Exception as e: + if debug:traceback.print_exc() + return "","异常信息:"+str(e)+"\n请排障后重试" + def modelUnload(): + global model + if model is None: + return sid.update(choices = [],value=""),"没有模型需要卸载!" + else: + model = None + torch.cuda.empty_cache() + return sid.update(choices = [],value=""),"模型卸载完毕!" + vc_submit.click(vc_fn, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,F0_mean_pooling,enhancer_adaptive_key], [vc_output1, vc_output2]) + vc_submit2.click(vc_fn2, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale,pad_seconds,cl_num,lg_num,lgr_num,text2tts,tts_rate,F0_mean_pooling,enhancer_adaptive_key], [vc_output1, vc_output2]) + model_analysis_button.click(modelAnalysis,[model_path,config_path,cluster_model_path,device,enhance],[sid,sid_output]) + model_unload_button.click(modelUnload,[],[sid,sid_output]) + app.launch() + +