| | |
| | | #include <stdio.h> |
| | | #include <stdlib.h> |
| | | #include <string.h> |
| | | #include <fstream> |
| | | #include <assert.h> |
| | | |
| | | #include "Audio.h" |
| | | #include "precomp.h" |
| | | |
| | | using namespace std; |
| | | |
| | | // see http://soundfile.sapp.org/doc/WaveFormat/ |
| | | // Note: We assume little endian here |
| | | struct WaveHeader { |
| | | bool Validate() const { |
| | | // F F I R |
| | | if (chunk_id != 0x46464952) { |
| | | printf("Expected chunk_id RIFF. Given: 0x%08x\n", chunk_id); |
| | | return false; |
| | | } |
| | | // E V A W |
| | | if (format != 0x45564157) { |
| | | printf("Expected format WAVE. Given: 0x%08x\n", format); |
| | | return false; |
| | | } |
| | | |
| | | if (subchunk1_id != 0x20746d66) { |
| | | printf("Expected subchunk1_id 0x20746d66. Given: 0x%08x\n", |
| | | subchunk1_id); |
| | | return false; |
| | | } |
| | | |
| | | if (subchunk1_size != 16) { // 16 for PCM |
| | | printf("Expected subchunk1_size 16. Given: %d\n", |
| | | subchunk1_size); |
| | | return false; |
| | | } |
| | | |
| | | if (audio_format != 1) { // 1 for PCM |
| | | printf("Expected audio_format 1. Given: %d\n", audio_format); |
| | | return false; |
| | | } |
| | | |
| | | if (num_channels != 1) { // we support only single channel for now |
| | | printf("Expected single channel. Given: %d\n", num_channels); |
| | | return false; |
| | | } |
| | | if (byte_rate != (sample_rate * num_channels * bits_per_sample / 8)) { |
| | | return false; |
| | | } |
| | | |
| | | if (block_align != (num_channels * bits_per_sample / 8)) { |
| | | return false; |
| | | } |
| | | |
| | | if (bits_per_sample != 16) { // we support only 16 bits per sample |
| | | printf("Expected bits_per_sample 16. Given: %d\n", |
| | | bits_per_sample); |
| | | return false; |
| | | } |
| | | return true; |
| | | } |
| | | |
| | | // See https://en.wikipedia.org/wiki/WAV#Metadata and |
| | | // https://www.robotplanet.dk/audio/wav_meta_data/riff_mci.pdf |
| | | void SeekToDataChunk(std::istream &is) { |
| | | // a t a d |
| | | while (is && subchunk2_id != 0x61746164) { |
| | | // const char *p = reinterpret_cast<const char *>(&subchunk2_id); |
| | | // printf("Skip chunk (%x): %c%c%c%c of size: %d\n", subchunk2_id, p[0], |
| | | // p[1], p[2], p[3], subchunk2_size); |
| | | is.seekg(subchunk2_size, std::istream::cur); |
| | | is.read(reinterpret_cast<char *>(&subchunk2_id), sizeof(int32_t)); |
| | | is.read(reinterpret_cast<char *>(&subchunk2_size), sizeof(int32_t)); |
| | | } |
| | | } |
| | | |
| | | int32_t chunk_id; |
| | | int32_t chunk_size; |
| | | int32_t format; |
| | | int32_t subchunk1_id; |
| | | int32_t subchunk1_size; |
| | | int16_t audio_format; |
| | | int16_t num_channels; |
| | | int32_t sample_rate; |
| | | int32_t byte_rate; |
| | | int16_t block_align; |
| | | int16_t bits_per_sample; |
| | | int32_t subchunk2_id; // a tag of this chunk |
| | | int32_t subchunk2_size; // size of subchunk2 |
| | | }; |
| | | static_assert(sizeof(WaveHeader) == WAV_HEADER_SIZE, ""); |
| | | |
| | | class AudioWindow { |
| | | private: |
| | |
| | | float frame_length = 400; |
| | | float frame_shift = 160; |
| | | float num_new_samples = |
| | | ceil((num_samples - 400) / frame_shift) * frame_shift + frame_length; |
| | | ceil((num_samples - frame_length) / frame_shift) * frame_shift + frame_length; |
| | | |
| | | end = start + num_new_samples; |
| | | len = (int)num_new_samples; |
| | |
| | | |
| | | void Audio::disp() |
| | | { |
| | | printf("Audio time is %f s. len is %d\n", (float)speech_len / 16000, |
| | | printf("Audio time is %f s. len is %d\n", (float)speech_len / model_sample_rate, |
| | | speech_len); |
| | | } |
| | | |
| | | float Audio::get_time_len() |
| | | { |
| | | return (float)speech_len / 16000; |
| | | //speech_len); |
| | | return (float)speech_len / model_sample_rate; |
| | | } |
| | | |
| | | bool Audio::loadwav(const char *filename) |
| | | void Audio::wavResample(int32_t sampling_rate, const float *waveform, |
| | | int32_t n) |
| | | { |
| | | printf( |
| | | "Creating a resampler:\n" |
| | | " in_sample_rate: %d\n" |
| | | " output_sample_rate: %d\n", |
| | | sampling_rate, static_cast<int32_t>(model_sample_rate)); |
| | | float min_freq = |
| | | std::min<int32_t>(sampling_rate, model_sample_rate); |
| | | float lowpass_cutoff = 0.99 * 0.5 * min_freq; |
| | | |
| | | int32_t lowpass_filter_width = 6; |
| | | //FIXME |
| | | //auto resampler = new LinearResample( |
| | | // sampling_rate, model_sample_rate, lowpass_cutoff, lowpass_filter_width); |
| | | auto resampler = std::make_unique<LinearResample>( |
| | | sampling_rate, model_sample_rate, lowpass_cutoff, lowpass_filter_width); |
| | | std::vector<float> samples; |
| | | resampler->Resample(waveform, n, true, &samples); |
| | | //reset speech_data |
| | | speech_len = samples.size(); |
| | | if (speech_data != NULL) { |
| | | free(speech_data); |
| | | } |
| | | speech_data = (float*)malloc(sizeof(float) * speech_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_len); |
| | | copy(samples.begin(), samples.end(), speech_data); |
| | | } |
| | | |
| | | bool Audio::loadwav(const char *filename, int32_t* sampling_rate) |
| | | { |
| | | WaveHeader header; |
| | | if (speech_data != NULL) { |
| | | free(speech_data); |
| | | } |
| | | if (speech_buff != NULL) { |
| | | free(speech_buff); |
| | | } |
| | | |
| | | |
| | | offset = 0; |
| | | |
| | | FILE *fp; |
| | | fp = fopen(filename, "rb"); |
| | | if (fp == nullptr) |
| | | std::ifstream is(filename, std::ifstream::binary); |
| | | is.read(reinterpret_cast<char *>(&header), sizeof(header)); |
| | | if(!is){ |
| | | fprintf(stderr, "Failed to read %s\n", filename); |
| | | return false; |
| | | fseek(fp, 0, SEEK_END); /*定位到文件末尾*/ |
| | | uint32_t nFileLen = ftell(fp); /*得到文件大小*/ |
| | | fseek(fp, 44, SEEK_SET); /*跳过wav文件头*/ |
| | | |
| | | speech_len = (nFileLen - 44) / 2; |
| | | speech_align_len = (int)(ceil((float)speech_len / align_size) * align_size); |
| | | speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_align_len); |
| | | } |
| | | |
| | | *sampling_rate = header.sample_rate; |
| | | // header.subchunk2_size contains the number of bytes in the data. |
| | | // As we assume each sample contains two bytes, so it is divided by 2 here |
| | | speech_len = header.subchunk2_size / 2; |
| | | speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_len); |
| | | |
| | | if (speech_buff) |
| | | { |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_align_len); |
| | | int ret = fread(speech_buff, sizeof(int16_t), speech_len, fp); |
| | | fclose(fp); |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_len); |
| | | is.read(reinterpret_cast<char *>(speech_buff), header.subchunk2_size); |
| | | if (!is) { |
| | | fprintf(stderr, "Failed to read %s\n", filename); |
| | | return false; |
| | | } |
| | | speech_data = (float*)malloc(sizeof(float) * speech_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_len); |
| | | |
| | | speech_data = (float*)malloc(sizeof(float) * speech_align_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_align_len); |
| | | int i; |
| | | float scale = 1; |
| | | |
| | | if (data_type == 1) { |
| | | scale = 32768; |
| | | } |
| | | |
| | | for (i = 0; i < speech_len; i++) { |
| | | for (int32_t i = 0; i != speech_len; ++i) { |
| | | speech_data[i] = (float)speech_buff[i] / scale; |
| | | } |
| | | |
| | | //resample |
| | | if(*sampling_rate != model_sample_rate){ |
| | | wavResample(*sampling_rate, speech_data, speech_len); |
| | | } |
| | | |
| | | AudioFrame* frame = new AudioFrame(speech_len); |
| | | frame_queue.push(frame); |
| | | |
| | | |
| | | return true; |
| | | } |
| | |
| | | return false; |
| | | } |
| | | |
| | | |
| | | bool Audio::loadwav(const char* buf, int nFileLen) |
| | | bool Audio::loadwav(const char* buf, int nFileLen, int32_t* sampling_rate) |
| | | { |
| | | |
| | | |
| | | |
| | | WaveHeader header; |
| | | if (speech_data != NULL) { |
| | | free(speech_data); |
| | | } |
| | | if (speech_buff != NULL) { |
| | | free(speech_buff); |
| | | } |
| | | |
| | | offset = 0; |
| | | |
| | | size_t nOffset = 0; |
| | | std::memcpy(&header, buf, sizeof(header)); |
| | | |
| | | #define WAV_HEADER_SIZE 44 |
| | | |
| | | speech_len = (nFileLen - WAV_HEADER_SIZE) / 2; |
| | | speech_align_len = (int)(ceil((float)speech_len / align_size) * align_size); |
| | | speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_align_len); |
| | | *sampling_rate = header.sample_rate; |
| | | speech_len = header.subchunk2_size / 2; |
| | | speech_buff = (int16_t *)malloc(sizeof(int16_t) * speech_len); |
| | | if (speech_buff) |
| | | { |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_align_len); |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_len); |
| | | memcpy((void*)speech_buff, (const void*)(buf + WAV_HEADER_SIZE), speech_len * sizeof(int16_t)); |
| | | |
| | | speech_data = (float*)malloc(sizeof(float) * speech_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_len); |
| | | |
| | | speech_data = (float*)malloc(sizeof(float) * speech_align_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_align_len); |
| | | int i; |
| | | float scale = 1; |
| | | |
| | | if (data_type == 1) { |
| | | scale = 32768; |
| | | } |
| | | |
| | | for (i = 0; i < speech_len; i++) { |
| | | for (int32_t i = 0; i != speech_len; ++i) { |
| | | speech_data[i] = (float)speech_buff[i] / scale; |
| | | } |
| | | |
| | | //resample |
| | | if(*sampling_rate != model_sample_rate){ |
| | | wavResample(*sampling_rate, speech_data, speech_len); |
| | | } |
| | | |
| | | AudioFrame* frame = new AudioFrame(speech_len); |
| | | frame_queue.push(frame); |
| | | |
| | | return true; |
| | | } |
| | | else |
| | | return false; |
| | | |
| | | } |
| | | |
| | | |
| | | bool Audio::loadpcmwav(const char* buf, int nBufLen) |
| | | bool Audio::loadpcmwav(const char* buf, int nBufLen, int32_t* sampling_rate) |
| | | { |
| | | if (speech_data != NULL) { |
| | | free(speech_data); |
| | |
| | | } |
| | | offset = 0; |
| | | |
| | | size_t nOffset = 0; |
| | | |
| | | |
| | | |
| | | speech_len = nBufLen / 2; |
| | | speech_align_len = (int)(ceil((float)speech_len / align_size) * align_size); |
| | | speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_align_len); |
| | | speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len); |
| | | if (speech_buff) |
| | | { |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_align_len); |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_len); |
| | | memcpy((void*)speech_buff, (const void*)buf, speech_len * sizeof(int16_t)); |
| | | |
| | | speech_data = (float*)malloc(sizeof(float) * speech_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_len); |
| | | |
| | | speech_data = (float*)malloc(sizeof(float) * speech_align_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_align_len); |
| | | |
| | | |
| | | int i; |
| | | float scale = 1; |
| | | |
| | | if (data_type == 1) { |
| | | scale = 32768; |
| | | } |
| | | |
| | | for (i = 0; i < speech_len; i++) { |
| | | for (int32_t i = 0; i != speech_len; ++i) { |
| | | speech_data[i] = (float)speech_buff[i] / scale; |
| | | } |
| | | |
| | | //resample |
| | | if(*sampling_rate != model_sample_rate){ |
| | | wavResample(*sampling_rate, speech_data, speech_len); |
| | | } |
| | | |
| | | AudioFrame* frame = new AudioFrame(speech_len); |
| | |
| | | } |
| | | else |
| | | return false; |
| | | |
| | | |
| | | } |
| | | |
| | | bool Audio::loadpcmwav(const char* filename) |
| | | bool Audio::loadpcmwav(const char* filename, int32_t* sampling_rate) |
| | | { |
| | | |
| | | if (speech_data != NULL) { |
| | | free(speech_data); |
| | | } |
| | |
| | | fseek(fp, 0, SEEK_SET); |
| | | |
| | | speech_len = (nFileLen) / 2; |
| | | speech_align_len = (int)(ceil((float)speech_len / align_size) * align_size); |
| | | speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_align_len); |
| | | speech_buff = (int16_t*)malloc(sizeof(int16_t) * speech_len); |
| | | if (speech_buff) |
| | | { |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_align_len); |
| | | memset(speech_buff, 0, sizeof(int16_t) * speech_len); |
| | | int ret = fread(speech_buff, sizeof(int16_t), speech_len, fp); |
| | | fclose(fp); |
| | | |
| | | speech_data = (float*)malloc(sizeof(float) * speech_align_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_align_len); |
| | | speech_data = (float*)malloc(sizeof(float) * speech_len); |
| | | memset(speech_data, 0, sizeof(float) * speech_len); |
| | | |
| | | |
| | | |
| | | int i; |
| | | float scale = 1; |
| | | |
| | | if (data_type == 1) { |
| | | scale = 32768; |
| | | } |
| | | |
| | | for (i = 0; i < speech_len; i++) { |
| | | for (int32_t i = 0; i != speech_len; ++i) { |
| | | speech_data[i] = (float)speech_buff[i] / scale; |
| | | } |
| | | |
| | | //resample |
| | | if(*sampling_rate != model_sample_rate){ |
| | | wavResample(*sampling_rate, speech_data, speech_len); |
| | | } |
| | | |
| | | AudioFrame* frame = new AudioFrame(speech_len); |
| | | frame_queue.push(frame); |
| | | |
| | | |
| | | return true; |
| | | } |
| | |
| | | return false; |
| | | |
| | | } |
| | | |
| | | |
| | | int Audio::fetch_chunck(float *&dout, int len) |
| | | { |