You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
705 lines
24 KiB
705 lines
24 KiB
// Real-time speech recognition of input from a microphone
|
|
//
|
|
// A very quick-n-dirty implementation serving mainly as a proof of concept.
|
|
//
|
|
|
|
#include "whisper.h"
|
|
|
|
#include <SDL.h>
|
|
#include <SDL_audio.h>
|
|
|
|
#include <cassert>
|
|
#include <cstdio>
|
|
#include <string>
|
|
#include <thread>
|
|
#include <vector>
|
|
#include <fstream>
|
|
#include <mutex>
|
|
|
|
// 500 -> 00:05.000
|
|
// 6000 -> 01:00.000
|
|
std::string to_timestamp(int64_t t) {
|
|
int64_t sec = t/100;
|
|
int64_t msec = t - sec*100;
|
|
int64_t min = sec/60;
|
|
sec = sec - min*60;
|
|
|
|
char buf[32];
|
|
snprintf(buf, sizeof(buf), "%02d:%02d.%03d", (int) min, (int) sec, (int) msec);
|
|
|
|
return std::string(buf);
|
|
}
|
|
|
|
// command-line parameters
|
|
struct whisper_params {
|
|
int32_t n_threads = std::min(4, (int32_t) std::thread::hardware_concurrency());
|
|
int32_t step_ms = 3000;
|
|
int32_t length_ms = 10000;
|
|
int32_t keep_ms = 200;
|
|
int32_t capture_id = -1;
|
|
int32_t max_tokens = 32;
|
|
int32_t audio_ctx = 0;
|
|
|
|
float vad_thold = 0.6f;
|
|
float freq_thold = 100.0f;
|
|
|
|
bool speed_up = false;
|
|
bool translate = false;
|
|
bool print_special = false;
|
|
bool no_context = true;
|
|
bool no_timestamps = false;
|
|
|
|
std::string language = "en";
|
|
std::string model = "models/ggml-base.en.bin";
|
|
std::string fname_out;
|
|
};
|
|
|
|
void whisper_print_usage(int argc, char ** argv, const whisper_params & params);
|
|
|
|
bool whisper_params_parse(int argc, char ** argv, whisper_params & params) {
|
|
for (int i = 1; i < argc; i++) {
|
|
std::string arg = argv[i];
|
|
|
|
if (arg == "-h" || arg == "--help") {
|
|
whisper_print_usage(argc, argv, params);
|
|
exit(0);
|
|
}
|
|
else if (arg == "-t" || arg == "--threads") { params.n_threads = std::stoi(argv[++i]); }
|
|
else if ( arg == "--step") { params.step_ms = std::stoi(argv[++i]); }
|
|
else if ( arg == "--length") { params.length_ms = std::stoi(argv[++i]); }
|
|
else if ( arg == "--keep") { params.keep_ms = std::stoi(argv[++i]); }
|
|
else if (arg == "-c" || arg == "--capture") { params.capture_id = std::stoi(argv[++i]); }
|
|
else if (arg == "-mt" || arg == "--max-tokens") { params.max_tokens = std::stoi(argv[++i]); }
|
|
else if (arg == "-ac" || arg == "--audio-ctx") { params.audio_ctx = std::stoi(argv[++i]); }
|
|
else if (arg == "-vth" || arg == "--vad-thold") { params.vad_thold = std::stof(argv[++i]); }
|
|
else if (arg == "-fth" || arg == "--freq-thold") { params.freq_thold = std::stof(argv[++i]); }
|
|
else if (arg == "-su" || arg == "--speed-up") { params.speed_up = true; }
|
|
else if (arg == "-tr" || arg == "--translate") { params.translate = true; }
|
|
else if (arg == "-ps" || arg == "--print-special") { params.print_special = true; }
|
|
else if (arg == "-kc" || arg == "--keep-context") { params.no_context = false; }
|
|
else if (arg == "-l" || arg == "--language") { params.language = argv[++i]; }
|
|
else if (arg == "-m" || arg == "--model") { params.model = argv[++i]; }
|
|
else if (arg == "-f" || arg == "--file") { params.fname_out = argv[++i]; }
|
|
else {
|
|
fprintf(stderr, "error: unknown argument: %s\n", arg.c_str());
|
|
whisper_print_usage(argc, argv, params);
|
|
exit(0);
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
void whisper_print_usage(int /*argc*/, char ** argv, const whisper_params & params) {
|
|
fprintf(stderr, "\n");
|
|
fprintf(stderr, "usage: %s [options]\n", argv[0]);
|
|
fprintf(stderr, "\n");
|
|
fprintf(stderr, "options:\n");
|
|
fprintf(stderr, " -h, --help [default] show this help message and exit\n");
|
|
fprintf(stderr, " -t N, --threads N [%-7d] number of threads to use during computation\n", params.n_threads);
|
|
fprintf(stderr, " --step N [%-7d] audio step size in milliseconds\n", params.step_ms);
|
|
fprintf(stderr, " --length N [%-7d] audio length in milliseconds\n", params.length_ms);
|
|
fprintf(stderr, " --keep N [%-7d] audio to keep from previous step in ms\n", params.keep_ms);
|
|
fprintf(stderr, " -c ID, --capture ID [%-7d] capture device ID\n", params.capture_id);
|
|
fprintf(stderr, " -mt N, --max-tokens N [%-7d] maximum number of tokens per audio chunk\n", params.max_tokens);
|
|
fprintf(stderr, " -ac N, --audio-ctx N [%-7d] audio context size (0 - all)\n", params.audio_ctx);
|
|
fprintf(stderr, " -vth N, --vad-thold N [%-7.2f] voice activity detection threshold\n", params.vad_thold);
|
|
fprintf(stderr, " -fth N, --freq-thold N [%-7.2f] high-pass frequency cutoff\n", params.freq_thold);
|
|
fprintf(stderr, " -su, --speed-up [%-7s] speed up audio by x2 (reduced accuracy)\n", params.speed_up ? "true" : "false");
|
|
fprintf(stderr, " -tr, --translate [%-7s] translate from source language to english\n", params.translate ? "true" : "false");
|
|
fprintf(stderr, " -ps, --print-special [%-7s] print special tokens\n", params.print_special ? "true" : "false");
|
|
fprintf(stderr, " -kc, --keep-context [%-7s] keep context between audio chunks\n", params.no_context ? "false" : "true");
|
|
fprintf(stderr, " -l LANG, --language LANG [%-7s] spoken language\n", params.language.c_str());
|
|
fprintf(stderr, " -m FNAME, --model FNAME [%-7s] model path\n", params.model.c_str());
|
|
fprintf(stderr, " -f FNAME, --file FNAME [%-7s] text output file name\n", params.fname_out.c_str());
|
|
fprintf(stderr, "\n");
|
|
}
|
|
|
|
//
|
|
// SDL Audio capture
|
|
//
|
|
|
|
class audio_async {
|
|
public:
|
|
audio_async(int len_ms);
|
|
~audio_async();
|
|
|
|
bool init(int capture_id, int sample_rate);
|
|
|
|
// start capturing audio via the provided SDL callback
|
|
// keep last len_ms seconds of audio in a circular buffer
|
|
bool resume();
|
|
bool pause();
|
|
bool clear();
|
|
|
|
// callback to be called by SDL
|
|
void callback(uint8_t * stream, int len);
|
|
|
|
// get audio data from the circular buffer
|
|
void get(int ms, std::vector<float> & audio);
|
|
|
|
private:
|
|
SDL_AudioDeviceID m_dev_id_in = 0;
|
|
|
|
int m_len_ms = 0;
|
|
int m_sample_rate = 0;
|
|
|
|
bool m_running = false;
|
|
std::mutex m_mutex;
|
|
|
|
std::vector<float> m_audio;
|
|
std::vector<float> m_audio_new;
|
|
size_t m_audio_pos = 0;
|
|
size_t m_audio_len = 0;
|
|
};
|
|
|
|
audio_async::audio_async(int len_ms) {
|
|
m_len_ms = len_ms;
|
|
}
|
|
|
|
audio_async::~audio_async() {
|
|
if (m_dev_id_in) {
|
|
SDL_CloseAudioDevice(m_dev_id_in);
|
|
}
|
|
}
|
|
|
|
bool audio_async::init(int capture_id, int sample_rate) {
|
|
SDL_LogSetPriority(SDL_LOG_CATEGORY_APPLICATION, SDL_LOG_PRIORITY_INFO);
|
|
|
|
if (SDL_Init(SDL_INIT_AUDIO) < 0) {
|
|
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "Couldn't initialize SDL: %s\n", SDL_GetError());
|
|
return false;
|
|
}
|
|
|
|
SDL_SetHintWithPriority(SDL_HINT_AUDIO_RESAMPLING_MODE, "medium", SDL_HINT_OVERRIDE);
|
|
|
|
{
|
|
int nDevices = SDL_GetNumAudioDevices(SDL_TRUE);
|
|
fprintf(stderr, "%s: found %d capture devices:\n", __func__, nDevices);
|
|
for (int i = 0; i < nDevices; i++) {
|
|
fprintf(stderr, "%s: - Capture device #%d: '%s'\n", __func__, i, SDL_GetAudioDeviceName(i, SDL_TRUE));
|
|
}
|
|
}
|
|
|
|
SDL_AudioSpec capture_spec_requested;
|
|
SDL_AudioSpec capture_spec_obtained;
|
|
|
|
SDL_zero(capture_spec_requested);
|
|
SDL_zero(capture_spec_obtained);
|
|
|
|
capture_spec_requested.freq = sample_rate;
|
|
capture_spec_requested.format = AUDIO_F32;
|
|
capture_spec_requested.channels = 1;
|
|
capture_spec_requested.samples = 1024;
|
|
capture_spec_requested.callback = [](void * userdata, uint8_t * stream, int len) {
|
|
audio_async * audio = (audio_async *) userdata;
|
|
audio->callback(stream, len);
|
|
};
|
|
capture_spec_requested.userdata = this;
|
|
|
|
if (capture_id >= 0) {
|
|
fprintf(stderr, "%s: attempt to open capture device %d : '%s' ...\n", __func__, capture_id, SDL_GetAudioDeviceName(capture_id, SDL_TRUE));
|
|
m_dev_id_in = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(capture_id, SDL_TRUE), SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
|
|
} else {
|
|
fprintf(stderr, "%s: attempt to open default capture device ...\n", __func__);
|
|
m_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
|
|
}
|
|
|
|
if (!m_dev_id_in) {
|
|
fprintf(stderr, "%s: couldn't open an audio device for capture: %s!\n", __func__, SDL_GetError());
|
|
m_dev_id_in = 0;
|
|
|
|
return false;
|
|
} else {
|
|
fprintf(stderr, "%s: obtained spec for input device (SDL Id = %d):\n", __func__, m_dev_id_in);
|
|
fprintf(stderr, "%s: - sample rate: %d\n", __func__, capture_spec_obtained.freq);
|
|
fprintf(stderr, "%s: - format: %d (required: %d)\n", __func__, capture_spec_obtained.format,
|
|
capture_spec_requested.format);
|
|
fprintf(stderr, "%s: - channels: %d (required: %d)\n", __func__, capture_spec_obtained.channels,
|
|
capture_spec_requested.channels);
|
|
fprintf(stderr, "%s: - samples per frame: %d\n", __func__, capture_spec_obtained.samples);
|
|
}
|
|
|
|
m_sample_rate = capture_spec_obtained.freq;
|
|
|
|
m_audio.resize((m_sample_rate*m_len_ms)/1000);
|
|
|
|
return true;
|
|
}
|
|
|
|
bool audio_async::resume() {
|
|
if (!m_dev_id_in) {
|
|
fprintf(stderr, "%s: no audio device to resume!\n", __func__);
|
|
return false;
|
|
}
|
|
|
|
if (m_running) {
|
|
fprintf(stderr, "%s: already running!\n", __func__);
|
|
return false;
|
|
}
|
|
|
|
SDL_PauseAudioDevice(m_dev_id_in, 0);
|
|
|
|
m_running = true;
|
|
|
|
return true;
|
|
}
|
|
|
|
bool audio_async::pause() {
|
|
if (!m_dev_id_in) {
|
|
fprintf(stderr, "%s: no audio device to pause!\n", __func__);
|
|
return false;
|
|
}
|
|
|
|
if (!m_running) {
|
|
fprintf(stderr, "%s: already paused!\n", __func__);
|
|
return false;
|
|
}
|
|
|
|
SDL_PauseAudioDevice(m_dev_id_in, 1);
|
|
|
|
m_running = false;
|
|
|
|
return true;
|
|
}
|
|
|
|
bool audio_async::clear() {
|
|
if (!m_dev_id_in) {
|
|
fprintf(stderr, "%s: no audio device to clear!\n", __func__);
|
|
return false;
|
|
}
|
|
|
|
if (!m_running) {
|
|
fprintf(stderr, "%s: not running!\n", __func__);
|
|
return false;
|
|
}
|
|
|
|
{
|
|
std::lock_guard<std::mutex> lock(m_mutex);
|
|
|
|
m_audio_pos = 0;
|
|
m_audio_len = 0;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
// callback to be called by SDL
|
|
void audio_async::callback(uint8_t * stream, int len) {
|
|
if (!m_running) {
|
|
return;
|
|
}
|
|
|
|
const size_t n_samples = len / sizeof(float);
|
|
|
|
m_audio_new.resize(n_samples);
|
|
memcpy(m_audio_new.data(), stream, n_samples * sizeof(float));
|
|
|
|
//fprintf(stderr, "%s: %zu samples, pos %zu, len %zu\n", __func__, n_samples, m_audio_pos, m_audio_len);
|
|
|
|
{
|
|
std::lock_guard<std::mutex> lock(m_mutex);
|
|
|
|
if (m_audio_pos + n_samples > m_audio.size()) {
|
|
const size_t n0 = m_audio.size() - m_audio_pos;
|
|
|
|
memcpy(&m_audio[m_audio_pos], stream, n0 * sizeof(float));
|
|
memcpy(&m_audio[0], &stream[n0], (n_samples - n0) * sizeof(float));
|
|
|
|
m_audio_pos = (m_audio_pos + n_samples) % m_audio.size();
|
|
m_audio_len = m_audio.size();
|
|
} else {
|
|
memcpy(&m_audio[m_audio_pos], stream, n_samples * sizeof(float));
|
|
|
|
m_audio_pos = (m_audio_pos + n_samples) % m_audio.size();
|
|
m_audio_len = std::min(m_audio_len + n_samples, m_audio.size());
|
|
}
|
|
}
|
|
}
|
|
|
|
void audio_async::get(int ms, std::vector<float> & result) {
|
|
if (!m_dev_id_in) {
|
|
fprintf(stderr, "%s: no audio device to get audio from!\n", __func__);
|
|
return;
|
|
}
|
|
|
|
if (!m_running) {
|
|
fprintf(stderr, "%s: not running!\n", __func__);
|
|
return;
|
|
}
|
|
|
|
result.clear();
|
|
|
|
{
|
|
std::lock_guard<std::mutex> lock(m_mutex);
|
|
|
|
if (ms <= 0) {
|
|
ms = m_len_ms;
|
|
}
|
|
|
|
size_t n_samples = (m_sample_rate * ms) / 1000;
|
|
if (n_samples > m_audio_len) {
|
|
n_samples = m_audio_len;
|
|
}
|
|
|
|
result.resize(n_samples);
|
|
|
|
int s0 = m_audio_pos - n_samples;
|
|
if (s0 < 0) {
|
|
s0 += m_audio.size();
|
|
}
|
|
|
|
if (s0 + n_samples > m_audio.size()) {
|
|
const size_t n0 = m_audio.size() - s0;
|
|
|
|
memcpy(result.data(), &m_audio[s0], n0 * sizeof(float));
|
|
memcpy(&result[n0], &m_audio[0], (n_samples - n0) * sizeof(float));
|
|
} else {
|
|
memcpy(result.data(), &m_audio[s0], n_samples * sizeof(float));
|
|
}
|
|
}
|
|
}
|
|
|
|
///////////////////////////
|
|
|
|
void high_pass_filter(std::vector<float> & data, float cutoff, float sample_rate) {
|
|
const float rc = 1.0f / (2.0f * M_PI * cutoff);
|
|
const float dt = 1.0f / sample_rate;
|
|
const float alpha = dt / (rc + dt);
|
|
|
|
float y = data[0];
|
|
|
|
for (size_t i = 1; i < data.size(); i++) {
|
|
y = alpha * (y + data[i] - data[i - 1]);
|
|
data[i] = y;
|
|
}
|
|
}
|
|
|
|
bool vad_simple(std::vector<float> & pcmf32, int sample_rate, int last_ms, float vad_thold, float freq_thold, bool verbose) {
|
|
const int n_samples = pcmf32.size();
|
|
const int n_samples_last = (sample_rate * last_ms) / 1000;
|
|
|
|
if (n_samples_last >= n_samples) {
|
|
// not enough samples - assume no speech
|
|
return false;
|
|
}
|
|
|
|
if (freq_thold > 0.0f) {
|
|
high_pass_filter(pcmf32, freq_thold, sample_rate);
|
|
}
|
|
|
|
float energy_all = 0.0f;
|
|
float energy_last = 0.0f;
|
|
|
|
for (int i = 0; i < n_samples; i++) {
|
|
energy_all += fabsf(pcmf32[i]);
|
|
|
|
if (i >= n_samples - n_samples_last) {
|
|
energy_last += fabsf(pcmf32[i]);
|
|
}
|
|
}
|
|
|
|
energy_all /= n_samples;
|
|
energy_last /= n_samples_last;
|
|
|
|
if (verbose) {
|
|
fprintf(stderr, "%s: energy_all: %f, energy_last: %f, vad_thold: %f, freq_thold: %f\n", __func__, energy_all, energy_last, vad_thold, freq_thold);
|
|
}
|
|
|
|
if (energy_last > vad_thold*energy_all) {
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
int main(int argc, char ** argv) {
|
|
whisper_params params;
|
|
|
|
if (whisper_params_parse(argc, argv, params) == false) {
|
|
return 1;
|
|
}
|
|
|
|
params.keep_ms = std::min(params.keep_ms, params.step_ms); // cannot be more than step_ms
|
|
|
|
const int n_samples_step = (params.step_ms *1e-3)*WHISPER_SAMPLE_RATE;
|
|
const int n_samples_len = (params.length_ms*1e-3)*WHISPER_SAMPLE_RATE;
|
|
const int n_samples_keep = (params.keep_ms *1e-3)*WHISPER_SAMPLE_RATE;
|
|
const int n_samples_30s = (30000 *1e-3)*WHISPER_SAMPLE_RATE;
|
|
|
|
const int n_new_line = params.length_ms / params.step_ms - 1; // number of steps to print new line
|
|
|
|
const bool use_vad = n_samples_step <= 0; // sliding window mode uses VAD
|
|
|
|
params.no_timestamps = !use_vad;
|
|
params.no_context = use_vad;
|
|
params.max_tokens = 0;
|
|
|
|
// init audio
|
|
|
|
audio_async audio(params.length_ms);
|
|
if (!audio.init(params.capture_id, WHISPER_SAMPLE_RATE)) {
|
|
fprintf(stderr, "%s: audio.init() failed!\n", __func__);
|
|
return 1;
|
|
}
|
|
|
|
audio.resume();
|
|
|
|
// whisper init
|
|
|
|
if (whisper_lang_id(params.language.c_str()) == -1) {
|
|
fprintf(stderr, "error: unknown language '%s'\n", params.language.c_str());
|
|
whisper_print_usage(argc, argv, params);
|
|
exit(0);
|
|
}
|
|
|
|
struct whisper_context * ctx = whisper_init(params.model.c_str());
|
|
|
|
std::vector<float> pcmf32 (n_samples_30s, 0.0f);
|
|
std::vector<float> pcmf32_old(n_samples_30s, 0.0f);
|
|
std::vector<float> pcmf32_new(n_samples_30s, 0.0f);
|
|
|
|
std::vector<whisper_token> prompt_tokens;
|
|
|
|
// print some info about the processing
|
|
{
|
|
fprintf(stderr, "\n");
|
|
if (!whisper_is_multilingual(ctx)) {
|
|
if (params.language != "en" || params.translate) {
|
|
params.language = "en";
|
|
params.translate = false;
|
|
fprintf(stderr, "%s: WARNING: model is not multilingual, ignoring language and translation options\n", __func__);
|
|
}
|
|
}
|
|
fprintf(stderr, "%s: processing %d samples (step = %.1f sec / len = %.1f sec / keep = %.1f sec), %d threads, lang = %s, task = %s, timestamps = %d ...\n",
|
|
__func__,
|
|
n_samples_step,
|
|
float(n_samples_step)/WHISPER_SAMPLE_RATE,
|
|
float(n_samples_len )/WHISPER_SAMPLE_RATE,
|
|
float(n_samples_keep)/WHISPER_SAMPLE_RATE,
|
|
params.n_threads,
|
|
params.language.c_str(),
|
|
params.translate ? "translate" : "transcribe",
|
|
params.no_timestamps ? 0 : 1);
|
|
|
|
if (!use_vad) {
|
|
fprintf(stderr, "%s: n_new_line = %d\n", __func__, n_new_line);
|
|
} else {
|
|
fprintf(stderr, "%s: using VAD, will transcribe on speech activity\n", __func__);
|
|
}
|
|
|
|
fprintf(stderr, "\n");
|
|
}
|
|
|
|
int n_iter = 0;
|
|
|
|
bool is_running = true;
|
|
|
|
std::ofstream fout;
|
|
if (params.fname_out.length() > 0) {
|
|
fout.open(params.fname_out);
|
|
if (!fout.is_open()) {
|
|
fprintf(stderr, "%s: failed to open output file '%s'!\n", __func__, params.fname_out.c_str());
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
printf("[Start speaking]");
|
|
fflush(stdout);
|
|
|
|
auto t_last = std::chrono::high_resolution_clock::now();
|
|
const auto t_start = t_last;
|
|
|
|
// main audio loop
|
|
while (is_running) {
|
|
// handle Ctrl + C
|
|
{
|
|
SDL_Event event;
|
|
while (SDL_PollEvent(&event)) {
|
|
switch (event.type) {
|
|
case SDL_QUIT:
|
|
{
|
|
is_running = false;
|
|
} break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!is_running) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!is_running) {
|
|
break;
|
|
}
|
|
|
|
// process new audio
|
|
|
|
if (!use_vad) {
|
|
while (true) {
|
|
audio.get(params.step_ms, pcmf32_new);
|
|
|
|
if ((int) pcmf32_new.size() > 2*n_samples_step) {
|
|
fprintf(stderr, "\n\n%s: WARNING: cannot process audio fast enough, dropping audio ...\n\n", __func__);
|
|
audio.clear();
|
|
continue;
|
|
}
|
|
|
|
if ((int) pcmf32_new.size() >= n_samples_step) {
|
|
audio.clear();
|
|
break;
|
|
}
|
|
|
|
SDL_Delay(1);
|
|
}
|
|
|
|
const int n_samples_new = pcmf32_new.size();
|
|
|
|
// take up to params.length_ms audio from previous iteration
|
|
const int n_samples_take = std::min((int) pcmf32_old.size(), std::max(0, n_samples_keep + n_samples_len - n_samples_new));
|
|
|
|
//printf("processing: take = %d, new = %d, old = %d\n", n_samples_take, n_samples_new, (int) pcmf32_old.size());
|
|
|
|
pcmf32.resize(n_samples_new + n_samples_take);
|
|
|
|
for (int i = 0; i < n_samples_take; i++) {
|
|
pcmf32[i] = pcmf32_old[pcmf32_old.size() - n_samples_take + i];
|
|
}
|
|
|
|
memcpy(pcmf32.data() + n_samples_take, pcmf32_new.data(), n_samples_new*sizeof(float));
|
|
|
|
pcmf32_old = pcmf32;
|
|
} else {
|
|
const auto t_now = std::chrono::high_resolution_clock::now();
|
|
const auto t_diff = std::chrono::duration_cast<std::chrono::milliseconds>(t_now - t_last).count();
|
|
|
|
if (t_diff < 2000) {
|
|
std::this_thread::sleep_for(std::chrono::milliseconds(100));
|
|
|
|
continue;
|
|
}
|
|
|
|
audio.get(2000, pcmf32_new);
|
|
|
|
if (vad_simple(pcmf32_new, WHISPER_SAMPLE_RATE, 1000, params.vad_thold, params.freq_thold, false)) {
|
|
audio.get(params.length_ms, pcmf32);
|
|
} else {
|
|
std::this_thread::sleep_for(std::chrono::milliseconds(100));
|
|
|
|
continue;
|
|
}
|
|
|
|
t_last = t_now;
|
|
}
|
|
|
|
// run the inference
|
|
{
|
|
whisper_full_params wparams = whisper_full_default_params(WHISPER_SAMPLING_GREEDY);
|
|
|
|
wparams.print_progress = false;
|
|
wparams.print_special = params.print_special;
|
|
wparams.print_realtime = false;
|
|
wparams.print_timestamps = !params.no_timestamps;
|
|
wparams.translate = params.translate;
|
|
wparams.no_context = true;
|
|
wparams.single_segment = !use_vad;
|
|
wparams.max_tokens = params.max_tokens;
|
|
wparams.language = params.language.c_str();
|
|
wparams.n_threads = params.n_threads;
|
|
|
|
wparams.audio_ctx = params.audio_ctx;
|
|
wparams.speed_up = params.speed_up;
|
|
|
|
wparams.prompt_tokens = params.no_context ? nullptr : prompt_tokens.data();
|
|
wparams.prompt_n_tokens = params.no_context ? 0 : prompt_tokens.size();
|
|
|
|
if (whisper_full(ctx, wparams, pcmf32.data(), pcmf32.size()) != 0) {
|
|
fprintf(stderr, "%s: failed to process audio\n", argv[0]);
|
|
return 6;
|
|
}
|
|
|
|
// print result;
|
|
{
|
|
if (!use_vad) {
|
|
printf("\33[2K\r");
|
|
|
|
// print long empty line to clear the previous line
|
|
printf("%s", std::string(100, ' ').c_str());
|
|
|
|
printf("\33[2K\r");
|
|
} else {
|
|
const int64_t t1 = (t_last - t_start).count()/1000000;
|
|
const int64_t t0 = std::max(0.0, t1 - pcmf32.size()*1000.0/WHISPER_SAMPLE_RATE);
|
|
|
|
printf("\n");
|
|
printf("### Transcription %d START | t0 = %d ms | t1 = %d ms\n", n_iter, (int) t0, (int) t1);
|
|
printf("\n");
|
|
}
|
|
|
|
const int n_segments = whisper_full_n_segments(ctx);
|
|
for (int i = 0; i < n_segments; ++i) {
|
|
const char * text = whisper_full_get_segment_text(ctx, i);
|
|
|
|
if (params.no_timestamps) {
|
|
printf("%s", text);
|
|
fflush(stdout);
|
|
|
|
if (params.fname_out.length() > 0) {
|
|
fout << text;
|
|
}
|
|
} else {
|
|
const int64_t t0 = whisper_full_get_segment_t0(ctx, i);
|
|
const int64_t t1 = whisper_full_get_segment_t1(ctx, i);
|
|
|
|
printf ("[%s --> %s] %s\n", to_timestamp(t0).c_str(), to_timestamp(t1).c_str(), text);
|
|
|
|
if (params.fname_out.length() > 0) {
|
|
fout << "[" << to_timestamp(t0) << " --> " << to_timestamp(t1) << "] " << text << std::endl;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (params.fname_out.length() > 0) {
|
|
fout << std::endl;
|
|
}
|
|
|
|
if (use_vad){
|
|
printf("\n");
|
|
printf("### Transcription %d END\n", n_iter);
|
|
}
|
|
}
|
|
|
|
++n_iter;
|
|
|
|
if (!use_vad && (n_iter % n_new_line) == 0) {
|
|
printf("\n");
|
|
|
|
// keep part of the audio for next iteration to try to mitigate word boundary issues
|
|
pcmf32_old = std::vector<float>(pcmf32.end() - n_samples_keep, pcmf32.end());
|
|
|
|
// Add tokens of the last full length segment as the prompt
|
|
if (!params.no_context) {
|
|
prompt_tokens.clear();
|
|
|
|
const int n_segments = whisper_full_n_segments(ctx);
|
|
for (int i = 0; i < n_segments; ++i) {
|
|
const int token_count = whisper_full_n_tokens(ctx, i);
|
|
for (int j = 0; j < token_count; ++j) {
|
|
prompt_tokens.push_back(whisper_full_get_token_id(ctx, i, j));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
audio.pause();
|
|
|
|
whisper_print_timings(ctx);
|
|
whisper_free(ctx);
|
|
|
|
return 0;
|
|
}
|