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| // Real-time speech recognition of input from a microphone | |
| // | |
| // A very quick-n-dirty implementation serving mainly as a proof of concept. | |
| // | |
| // 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 (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; | |
| 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; | |
| } | |