// 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"
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 no_context = true;
bool print_special = false;
- bool no_timestamps = true;
+ bool no_context = true;
+ bool no_timestamps = false;
std::string language = "en";
std::string model = "models/ggml-base.en.bin";
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 == "-kc" || arg == "--keep-context") { params.no_context = false; }
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]; }
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, " -kc, --keep-context [%-7s] keep context between audio chunks\n", params.no_context ? "false" : "true");
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());
// SDL Audio capture
//
-SDL_AudioDeviceID g_dev_id_in = 0;
+class audio_async {
+public:
+ audio_async(int len_ms);
+ ~audio_async();
-bool audio_sdl_init(const int capture_id) {
- if (g_dev_id_in) {
- fprintf(stderr, "%s: already initialized\n", __func__);
- return false;
+ 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 (1);
+ return false;
}
SDL_SetHintWithPriority(SDL_HINT_AUDIO_RESAMPLING_MODE, "medium", SDL_HINT_OVERRIDE);
SDL_zero(capture_spec_requested);
SDL_zero(capture_spec_obtained);
- capture_spec_requested.freq = WHISPER_SAMPLE_RATE;
+ 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));
- g_dev_id_in = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(capture_id, SDL_TRUE), SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
+ 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__);
- g_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
+ m_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
}
- if (!g_dev_id_in) {
+
+ if (!m_dev_id_in) {
fprintf(stderr, "%s: couldn't open an audio device for capture: %s!\n", __func__, SDL_GetError());
- g_dev_id_in = 0;
+ m_dev_id_in = 0;
+
+ return false;
} else {
- fprintf(stderr, "%s: obtained spec for input device (SDL Id = %d):\n", __func__, g_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);
+ 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 (size_t 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;
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;
+
+ params.no_timestamps = !use_vad;
+ params.no_context = use_vad;
+ params.max_tokens = 0;
+
// init audio
- if (!audio_sdl_init(params.capture_id)) {
- fprintf(stderr, "%s: audio_sdl_init() failed!\n", __func__);
+ 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);
}
- // whisper init
-
struct whisper_context * ctx = whisper_init(params.model.c_str());
- const int n_samples = (params.step_ms/1000.0)*WHISPER_SAMPLE_RATE;
- const int n_samples_len = (params.length_ms/1000.0)*WHISPER_SAMPLE_RATE;
- const int n_samples_30s = 30*WHISPER_SAMPLE_RATE;
- const int n_samples_keep = 0.2*WHISPER_SAMPLE_RATE;
-
- std::vector<float> pcmf32(n_samples_30s, 0.0f);
- std::vector<float> pcmf32_old;
+ 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;
- const int n_new_line = params.length_ms / params.step_ms - 1;
// print some info about the processing
{
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), %d threads, lang = %s, task = %s, timestamps = %d ...\n",
+ 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,
- float(n_samples)/WHISPER_SAMPLE_RATE,
- float(n_samples_len)/WHISPER_SAMPLE_RATE,
+ 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);
- fprintf(stderr, "%s: n_new_line = %d\n", __func__, n_new_line);
+ 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");
}
- SDL_PauseAudioDevice(g_dev_id_in, 0);
-
int n_iter = 0;
+
bool is_running = true;
std::ofstream fout;
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
}
// process new audio
- if (n_iter > 0 && SDL_GetQueuedAudioSize(g_dev_id_in) > 2*n_samples*sizeof(float)) {
- fprintf(stderr, "\n\n%s: WARNING: cannot process audio fast enough, dropping audio ...\n\n", __func__);
- SDL_ClearQueuedAudio(g_dev_id_in);
- }
- while (SDL_GetQueuedAudioSize(g_dev_id_in) < n_samples*sizeof(float)) {
- SDL_Delay(1);
- }
+ if (!use_vad) {
+ while (true) {
+ audio.get(params.step_ms, pcmf32_new);
- const int n_samples_new = SDL_GetQueuedAudioSize(g_dev_id_in)/sizeof(float);
+ 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;
+ }
- // take one second from previous iteration
- //const int n_samples_take = std::min((int) pcmf32_old.size(), std::max(0, n_samples_30s/30 - n_samples_new));
+ if ((int) pcmf32_new.size() >= n_samples_step) {
+ audio.clear();
+ break;
+ }
- // 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));
+ SDL_Delay(1);
+ }
- //printf("processing: take = %d, new = %d, old = %d\n", n_samples_take, n_samples_new, (int) pcmf32_old.size());
+ const int n_samples_new = pcmf32_new.size();
- pcmf32.resize(n_samples_new + n_samples_take);
+ // 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));
- for (int i = 0; i < n_samples_take; i++) {
- pcmf32[i] = pcmf32_old[pcmf32_old.size() - n_samples_take + i];
- }
+ //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;
+ }
- SDL_DequeueAudio(g_dev_id_in, pcmf32.data() + n_samples_take, n_samples_new*sizeof(float));
+ audio.get(2000, pcmf32_new);
- pcmf32_old = pcmf32;
+ 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
{
wparams.print_timestamps = !params.no_timestamps;
wparams.translate = params.translate;
wparams.no_context = true;
- wparams.single_segment = true;
+ wparams.single_segment = !use_vad;
wparams.max_tokens = params.max_tokens;
wparams.language = params.language.c_str();
wparams.n_threads = params.n_threads;
// print result;
{
- printf("\33[2K\r");
+ if (!use_vad) {
+ printf("\33[2K\r");
- // print long empty line to clear the previous line
- printf("%s", std::string(100, ' ').c_str());
+ // print long empty line to clear the previous line
+ printf("%s", std::string(100, ' ').c_str());
- printf("\33[2K\r");
+ 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 = %lld ms | t1 = %lld ms\n", n_iter, t0, t1);
+ printf("\n");
+ }
const int n_segments = whisper_full_n_segments(ctx);
for (int i = 0; i < n_segments; ++i) {
if (params.fname_out.length() > 0) {
fout << std::endl;
}
+
+ if (use_vad){
+ printf("\n");
+ printf("### Transcription %d END\n", n_iter);
+ }
}
++n_iter;
- if ((n_iter % n_new_line) == 0) {
+ 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
}
}
- if (g_dev_id_in >= 0) {
- SDL_CloseAudioDevice(g_dev_id_in);
- }
+ audio.pause();
whisper_print_timings(ctx);
whisper_free(ctx);
overview / pkg / ggml / sources / whisper.cpp / commitdiff