usage: ./main [options] file0.wav file1.wav ...
options:
- -h, --help show this help message and exit
- -t N, --threads N number of threads to use during computation (default: 4)
- -p N, --processors N number of processors to use during computation (default: 1)
- -ot N, --offset-t N time offset in milliseconds (default: 0)
- -on N, --offset-n N segment index offset (default: 0)
- -mc N, --max-context N maximum number of text context tokens to store (default: max)
- -ml N, --max-len N maximum segment length in characters (default: 0)
- -wt N, --word-thold N word timestamp probability threshold (default: 0.010000)
- -v, --verbose verbose output
- --translate translate from source language to english
- -otxt, --output-txt output result in a text file
- -ovtt, --output-vtt output result in a vtt file
- -osrt, --output-srt output result in a srt file
- -owts, --output-words output script for generating karaoke video
- -ps, --print_special print special tokens
- -pc, --print_colors print colors
- -nt, --no_timestamps do not print timestamps
- -l LANG, --language LANG spoken language (default: en)
- -m FNAME, --model FNAME model path (default: models/ggml-base.en.bin)
- -f FNAME, --file FNAME input WAV file path
+ -h, --help [default] show this help message and exit
+ -t N, --threads N [4 ] number of threads to use during computation
+ -p N, --processors N [1 ] number of processors to use during computation
+ -ot N, --offset-t N [0 ] time offset in milliseconds
+ -on N, --offset-n N [0 ] segment index offset
+ -d N, --duration N [0 ] duration of audio to process in milliseconds
+ -mc N, --max-context N [-1 ] maximum number of text context tokens to store
+ -ml N, --max-len N [0 ] maximum segment length in characters
+ -wt N, --word-thold N [0.01 ] word timestamp probability threshold
+ -su, --speed-up [false ] speed up audio by x2 (reduced accuracy)
+ -tr, --translate [false ] translate from source language to english
+ -otxt, --output-txt [false ] output result in a text file
+ -ovtt, --output-vtt [false ] output result in a vtt file
+ -osrt, --output-srt [false ] output result in a srt file
+ -owts, --output-words [false ] output script for generating karaoke video
+ -ps, --print-special [false ] print special tokens
+ -pc, --print-colors [false ] print colors
+ -nt, --no-timestamps [true ] do not print timestamps
+ -l LANG, --language LANG [en ] spoken language
+ -m FNAME, --model FNAME [models/ggml-base.en.bin] model path
+ -f FNAME, --file FNAME [ ] input WAV file path
bash ./models/download-ggml-model.sh base.en
Downloading ggml model base.en ...
whisper_model_load: n_mels = 80
whisper_model_load: f16 = 1
whisper_model_load: type = 2
-whisper_model_load: mem_required = 670.00 MB
whisper_model_load: adding 1607 extra tokens
-whisper_model_load: ggml ctx size = 140.60 MB
-whisper_model_load: memory size = 22.83 MB
-whisper_model_load: model size = 140.54 MB
+whisper_model_load: mem_required = 506.00 MB
+whisper_model_load: ggml ctx size = 140.60 MB
+whisper_model_load: memory size = 22.83 MB
+whisper_model_load: model size = 140.54 MB
-system_info: n_threads = 4 / 10 | AVX2 = 0 | AVX512 = 0 | NEON = 1 | FP16_VA = 1 | WASM_SIMD = 0 | BLAS = 1 |
+system_info: n_threads = 4 / 10 | AVX = 0 | AVX2 = 0 | AVX512 = 0 | NEON = 1 | FP16_VA = 1 | WASM_SIMD = 0 | BLAS = 1 |
main: processing 'samples/jfk.wav' (176000 samples, 11.0 sec), 4 threads, 1 processors, lang = en, task = transcribe, timestamps = 1 ...
--- /dev/null
+// Voice assistant example
+//
+// Speak short text commands to the microphone.
+// This program will detect your voice command and convert them to text.
+//
+// ref: https://github.com/ggerganov/whisper.cpp/issues/171
+//
+
+#include "whisper.h"
+
+#include <SDL.h>
+#include <SDL_audio.h>
+
+#include <cassert>
+#include <cstdio>
+#include <string>
+#include <thread>
+#include <vector>
+#include <fstream>
+#include <regex>
+
+// command-line parameters
+struct whisper_params {
+ int32_t n_threads = std::min(4, (int32_t) std::thread::hardware_concurrency());
+ int32_t prompt_ms = 5000;
+ int32_t command_ms = 4000;
+ 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 print_energy = false;
+ bool no_timestamps = true;
+
+ 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 == "-pms" || arg == "--prompt-ms") { params.prompt_ms = std::stoi(argv[++i]); }
+ else if (arg == "-cms" || arg == "--command-ms") { params.command_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 == "-pe" || arg == "--print-energy") { params.print_energy = true; }
+ 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, " -pms N, --prompt-ms N [%-7d] prompt duration in milliseconds\n", params.prompt_ms);
+ fprintf(stderr, " -cms N, --command-ms N [%-7d] command duration in milliseconds\n", params.command_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, " -pe, --print-energy [%-7s] print sound energy (for debugging)\n", params.print_energy ? "true" : "false");
+ 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) {
+ m_len_ms = len_ms;
+ }
+
+ 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;
+};
+
+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));
+ }
+ }
+}
+
+///////////////////////////
+
+std::string trim(const std::string & s) {
+ std::regex e("^\\s+|\\s+$");
+ return std::regex_replace(s, e, "");
+}
+
+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;
+}
+
+std::string transcribe(whisper_context * ctx, const whisper_params & params, const std::vector<float> & pcmf32, float & prob, int64_t & t_ms) {
+ const auto t_start = std::chrono::high_resolution_clock::now();
+
+ prob = 0.0f;
+ t_ms = 0;
+
+ 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 = true;
+ 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;
+
+ if (whisper_full(ctx, wparams, pcmf32.data(), pcmf32.size()) != 0) {
+ return "";
+ }
+
+ int prob_n = 0;
+ std::string result;
+
+ 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);
+
+ result += text;
+
+ const int n_tokens = whisper_full_n_tokens(ctx, i);
+ for (int j = 0; j < n_tokens; ++j) {
+ const auto token = whisper_full_get_token_data(ctx, i, j);
+
+ prob += token.p;
+ ++prob_n;
+ }
+ }
+
+ if (prob_n > 0) {
+ prob /= prob_n;
+ }
+
+ const auto t_end = std::chrono::high_resolution_clock::now();
+ t_ms = std::chrono::duration_cast<std::chrono::milliseconds>(t_end - t_start).count();
+
+ return result;
+}
+
+// compute similarity between two strings using Levenshtein distance
+float similarity(const std::string & s0, const std::string & s1) {
+ const size_t len0 = s0.size() + 1;
+ const size_t len1 = s1.size() + 1;
+
+ std::vector<int> col(len1, 0);
+ std::vector<int> prevCol(len1, 0);
+
+ for (size_t i = 0; i < len1; i++) {
+ prevCol[i] = i;
+ }
+
+ for (size_t i = 0; i < len0; i++) {
+ col[0] = i;
+ for (size_t j = 1; j < len1; j++) {
+ col[j] = std::min(std::min(1 + col[j - 1], 1 + prevCol[j]), prevCol[j - 1] + (s0[i - 1] == s1[j - 1] ? 0 : 1));
+ }
+ col.swap(prevCol);
+ }
+
+ const float dist = prevCol[len1 - 1];
+
+ return 1.0f - (dist / std::max(s0.size(), s1.size()));
+}
+
+int main(int argc, char ** argv) {
+ whisper_params params;
+
+ if (whisper_params_parse(argc, argv, params) == false) {
+ return 1;
+ }
+
+ 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());
+
+ // 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 threads, lang = %s, task = %s, timestamps = %d ...\n",
+ __func__,
+ params.n_threads,
+ params.language.c_str(),
+ params.translate ? "translate" : "transcribe",
+ params.no_timestamps ? 0 : 1);
+
+ fprintf(stderr, "\n");
+ }
+
+
+ // init audio
+
+ audio_async audio(30*1000);
+ if (!audio.init(params.capture_id, WHISPER_SAMPLE_RATE)) {
+ fprintf(stderr, "%s: audio.init() failed!\n", __func__);
+ return 1;
+ }
+
+ audio.resume();
+
+ bool is_running = true;
+ bool have_prompt = false;
+ bool ask_prompt = true;
+
+ float prob0 = 0.0f;
+ float prob = 0.0f;
+
+ std::vector<float> pcmf32_cur;
+ std::vector<float> pcmf32_prompt;
+
+ const std::string k_prompt = "Ok Whisper, start listening for commands.";
+
+ // main 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;
+ }
+ }
+
+ // delay
+ std::this_thread::sleep_for(std::chrono::milliseconds(100));
+
+ if (ask_prompt) {
+ fprintf(stdout, "\n");
+ fprintf(stdout, "%s: Say the following phrase: '%s'\n", __func__, k_prompt.c_str());
+ fprintf(stdout, "\n");
+
+ ask_prompt = false;
+ }
+
+ int64_t t_ms = 0;
+
+ {
+ audio.get(2000, pcmf32_cur);
+
+ if (vad_simple(pcmf32_cur, WHISPER_SAMPLE_RATE, 1000, params.vad_thold, params.freq_thold, params.print_energy)) {
+ fprintf(stdout, "%s: Speech detected!\n", __func__);
+
+ if (!have_prompt) {
+ audio.get(params.prompt_ms, pcmf32_cur);
+
+ const auto txt = ::trim(::transcribe(ctx, params, pcmf32_cur, prob0, t_ms));
+
+ fprintf(stdout, "%s: Heard '%s', (prob = %6.3f, t = %d ms)\n", __func__, txt.c_str(), prob0, (int) t_ms);
+
+ const float sim = similarity(txt, k_prompt);
+
+ if (txt.length() < 0.8*k_prompt.length() || txt.length() > 1.2*k_prompt.length() || sim < 0.8f) {
+ fprintf(stdout, "%s: WARNING: prompt not recognized, try again\n", __func__);
+ ask_prompt = true;
+ } else {
+ fprintf(stdout, "\n");
+ fprintf(stdout, "%s: The prompt has been recognized!\n", __func__);
+ fprintf(stdout, "%s: Waiting for voice commands ...\n", __func__);
+ fprintf(stdout, "\n");
+
+ // save the audio for the prompt
+ pcmf32_prompt = pcmf32_cur;
+ have_prompt = true;
+ }
+ } else {
+ audio.get(params.command_ms, pcmf32_cur);
+
+ // prepend the prompt audio
+ pcmf32_cur.insert(pcmf32_cur.begin(), pcmf32_prompt.begin(), pcmf32_prompt.end());
+
+ const auto txt = ::trim(::transcribe(ctx, params, pcmf32_cur, prob, t_ms));
+
+ printf("prob0 = %6.3f, prob = %6.3f, t = %d ms\n", prob0, prob, (int) t_ms);
+ prob = 100.0f*(prob - prob0);
+
+ //fprintf(stdout, "%s: heard '%s'\n", __func__, txt.c_str());
+
+ // find the prompt in the text
+ float best_sim = 0.0f;
+ size_t best_len = 0;
+ for (int n = 0.8*k_prompt.size(); n <= 1.2*k_prompt.size(); ++n) {
+ const auto prompt = txt.substr(0, n);
+
+ const float sim = similarity(prompt, k_prompt);
+
+ //fprintf(stderr, "%s: prompt = '%s', sim = %f\n", __func__, prompt.c_str(), sim);
+
+ if (sim > best_sim) {
+ best_sim = sim;
+ best_len = n;
+ }
+ }
+
+ const std::string command = ::trim(txt.substr(best_len));
+
+ fprintf(stdout, "%s: Command '%s', (prob = %6.3f, t = %d ms)\n", __func__, command.c_str(), prob, (int) t_ms);
+ fprintf(stdout, "\n");
+ }
+
+ audio.clear();
+ }
+ }
+ }
+
+ audio.pause();
+
+ whisper_print_timings(ctx);
+ whisper_free(ctx);
+
+ return 0;
+}
overview / pkg / ggml / sources / whisper.cpp / commitdiff