mirror of
https://github.com/ggerganov/whisper.cpp.git
synced 2024-12-18 20:27:53 +00:00
stream : add sliding window mode
This commit is contained in:
Georgi Gerganov
parent
124c718c73
commit
b0f8013eb9
@ -1,6 +1,7 @@
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// Real-time speech recognition of input from a microphone
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//
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// A very quick-n-dirty implementation serving mainly as a proof of concept.
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//
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#include "whisper.h"
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@ -33,15 +34,19 @@ struct whisper_params {
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int32_t n_threads = std::min(4, (int32_t) std::thread::hardware_concurrency());
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int32_t step_ms = 3000;
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int32_t length_ms = 10000;
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int32_t keep_ms = 200;
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int32_t capture_id = -1;
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int32_t max_tokens = 32;
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int32_t audio_ctx = 0;
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float vad_thold = 0.6f;
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float freq_thold = 100.0f;
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bool speed_up = false;
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bool translate = false;
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bool no_context = true;
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bool print_special = false;
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bool no_timestamps = true;
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bool no_context = true;
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bool no_timestamps = false;
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std::string language = "en";
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std::string model = "models/ggml-base.en.bin";
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@ -61,13 +66,16 @@ bool whisper_params_parse(int argc, char ** argv, whisper_params & params) {
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else if (arg == "-t" || arg == "--threads") { params.n_threads = std::stoi(argv[++i]); }
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else if ( arg == "--step") { params.step_ms = std::stoi(argv[++i]); }
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else if ( arg == "--length") { params.length_ms = std::stoi(argv[++i]); }
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else if ( arg == "--keep") { params.keep_ms = std::stoi(argv[++i]); }
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else if (arg == "-c" || arg == "--capture") { params.capture_id = std::stoi(argv[++i]); }
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else if (arg == "-mt" || arg == "--max-tokens") { params.max_tokens = std::stoi(argv[++i]); }
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else if (arg == "-ac" || arg == "--audio-ctx") { params.audio_ctx = std::stoi(argv[++i]); }
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else if (arg == "-vth" || arg == "--vad-thold") { params.vad_thold = std::stof(argv[++i]); }
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else if (arg == "-fth" || arg == "--freq-thold") { params.freq_thold = std::stof(argv[++i]); }
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else if (arg == "-su" || arg == "--speed-up") { params.speed_up = true; }
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else if (arg == "-tr" || arg == "--translate") { params.translate = true; }
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else if (arg == "-kc" || arg == "--keep-context") { params.no_context = false; }
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else if (arg == "-ps" || arg == "--print-special") { params.print_special = true; }
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else if (arg == "-kc" || arg == "--keep-context") { params.no_context = false; }
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else if (arg == "-l" || arg == "--language") { params.language = argv[++i]; }
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else if (arg == "-m" || arg == "--model") { params.model = argv[++i]; }
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else if (arg == "-f" || arg == "--file") { params.fname_out = argv[++i]; }
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@ -90,13 +98,16 @@ void whisper_print_usage(int argc, char ** argv, const whisper_params & params)
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fprintf(stderr, " -t N, --threads N [%-7d] number of threads to use during computation\n", params.n_threads);
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fprintf(stderr, " --step N [%-7d] audio step size in milliseconds\n", params.step_ms);
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fprintf(stderr, " --length N [%-7d] audio length in milliseconds\n", params.length_ms);
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fprintf(stderr, " --keep N [%-7d] audio to keep from previous step in ms\n", params.keep_ms);
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fprintf(stderr, " -c ID, --capture ID [%-7d] capture device ID\n", params.capture_id);
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fprintf(stderr, " -mt N, --max-tokens N [%-7d] maximum number of tokens per audio chunk\n", params.max_tokens);
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fprintf(stderr, " -ac N, --audio-ctx N [%-7d] audio context size (0 - all)\n", params.audio_ctx);
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fprintf(stderr, " -vth N, --vad-thold N [%-7.2f] voice activity detection threshold\n", params.vad_thold);
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fprintf(stderr, " -fth N, --freq-thold N [%-7.2f] high-pass frequency cutoff\n", params.freq_thold);
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fprintf(stderr, " -su, --speed-up [%-7s] speed up audio by x2 (reduced accuracy)\n", params.speed_up ? "true" : "false");
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fprintf(stderr, " -tr, --translate [%-7s] translate from source language to english\n", params.translate ? "true" : "false");
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fprintf(stderr, " -kc, --keep-context [%-7s] keep context between audio chunks\n", params.no_context ? "false" : "true");
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fprintf(stderr, " -ps, --print-special [%-7s] print special tokens\n", params.print_special ? "true" : "false");
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fprintf(stderr, " -kc, --keep-context [%-7s] keep context between audio chunks\n", params.no_context ? "false" : "true");
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fprintf(stderr, " -l LANG, --language LANG [%-7s] spoken language\n", params.language.c_str());
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fprintf(stderr, " -m FNAME, --model FNAME [%-7s] model path\n", params.model.c_str());
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fprintf(stderr, " -f FNAME, --file FNAME [%-7s] text output file name\n", params.fname_out.c_str());
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@ -107,19 +118,56 @@ void whisper_print_usage(int argc, char ** argv, const whisper_params & params)
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// SDL Audio capture
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//
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SDL_AudioDeviceID g_dev_id_in = 0;
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class audio_async {
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public:
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audio_async(int len_ms);
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~audio_async();
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bool audio_sdl_init(const int capture_id) {
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if (g_dev_id_in) {
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fprintf(stderr, "%s: already initialized\n", __func__);
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return false;
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bool init(int capture_id, int sample_rate);
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// start capturing audio via the provided SDL callback
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// keep last len_ms seconds of audio in a circular buffer
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bool resume();
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bool pause();
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bool clear();
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// callback to be called by SDL
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void callback(uint8_t * stream, int len);
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// get audio data from the circular buffer
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void get(int ms, std::vector<float> & audio);
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private:
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SDL_AudioDeviceID m_dev_id_in = 0;
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int m_len_ms = 0;
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int m_sample_rate = 0;
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bool m_running = false;
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std::mutex m_mutex;
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std::vector<float> m_audio;
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std::vector<float> m_audio_new;
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size_t m_audio_pos = 0;
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size_t m_audio_len = 0;
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};
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audio_async::audio_async(int len_ms) {
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m_len_ms = len_ms;
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}
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audio_async::~audio_async() {
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if (m_dev_id_in) {
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SDL_CloseAudioDevice(m_dev_id_in);
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}
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}
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bool audio_async::init(int capture_id, int sample_rate) {
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SDL_LogSetPriority(SDL_LOG_CATEGORY_APPLICATION, SDL_LOG_PRIORITY_INFO);
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if (SDL_Init(SDL_INIT_AUDIO) < 0) {
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SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "Couldn't initialize SDL: %s\n", SDL_GetError());
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return (1);
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return false;
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}
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SDL_SetHintWithPriority(SDL_HINT_AUDIO_RESAMPLING_MODE, "medium", SDL_HINT_OVERRIDE);
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@ -138,34 +186,232 @@ bool audio_sdl_init(const int capture_id) {
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SDL_zero(capture_spec_requested);
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SDL_zero(capture_spec_obtained);
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capture_spec_requested.freq = WHISPER_SAMPLE_RATE;
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capture_spec_requested.freq = sample_rate;
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capture_spec_requested.format = AUDIO_F32;
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capture_spec_requested.channels = 1;
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capture_spec_requested.samples = 1024;
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capture_spec_requested.callback = [](void * userdata, uint8_t * stream, int len) {
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audio_async * audio = (audio_async *) userdata;
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audio->callback(stream, len);
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};
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capture_spec_requested.userdata = this;
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if (capture_id >= 0) {
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fprintf(stderr, "%s: attempt to open capture device %d : '%s' ...\n", __func__, capture_id, SDL_GetAudioDeviceName(capture_id, SDL_TRUE));
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g_dev_id_in = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(capture_id, SDL_TRUE), SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
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m_dev_id_in = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(capture_id, SDL_TRUE), SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
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} else {
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fprintf(stderr, "%s: attempt to open default capture device ...\n", __func__);
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g_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
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m_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
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}
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if (!g_dev_id_in) {
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if (!m_dev_id_in) {
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fprintf(stderr, "%s: couldn't open an audio device for capture: %s!\n", __func__, SDL_GetError());
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g_dev_id_in = 0;
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m_dev_id_in = 0;
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return false;
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} else {
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fprintf(stderr, "%s: obtained spec for input device (SDL Id = %d):\n", __func__, g_dev_id_in);
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fprintf(stderr, "%s: - sample rate: %d\n", __func__, capture_spec_obtained.freq);
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fprintf(stderr, "%s: - format: %d (required: %d)\n", __func__, capture_spec_obtained.format, capture_spec_requested.format);
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fprintf(stderr, "%s: - channels: %d (required: %d)\n", __func__, capture_spec_obtained.channels, capture_spec_requested.channels);
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fprintf(stderr, "%s: - samples per frame: %d\n", __func__, capture_spec_obtained.samples);
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fprintf(stderr, "%s: obtained spec for input device (SDL Id = %d):\n", __func__, m_dev_id_in);
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fprintf(stderr, "%s: - sample rate: %d\n", __func__, capture_spec_obtained.freq);
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fprintf(stderr, "%s: - format: %d (required: %d)\n", __func__, capture_spec_obtained.format,
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capture_spec_requested.format);
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fprintf(stderr, "%s: - channels: %d (required: %d)\n", __func__, capture_spec_obtained.channels,
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capture_spec_requested.channels);
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fprintf(stderr, "%s: - samples per frame: %d\n", __func__, capture_spec_obtained.samples);
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}
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m_sample_rate = capture_spec_obtained.freq;
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m_audio.resize((m_sample_rate*m_len_ms)/1000);
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return true;
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}
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bool audio_async::resume() {
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if (!m_dev_id_in) {
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fprintf(stderr, "%s: no audio device to resume!\n", __func__);
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return false;
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}
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if (m_running) {
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fprintf(stderr, "%s: already running!\n", __func__);
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return false;
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}
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SDL_PauseAudioDevice(m_dev_id_in, 0);
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m_running = true;
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return true;
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}
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bool audio_async::pause() {
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if (!m_dev_id_in) {
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fprintf(stderr, "%s: no audio device to pause!\n", __func__);
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return false;
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}
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if (!m_running) {
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fprintf(stderr, "%s: already paused!\n", __func__);
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return false;
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}
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SDL_PauseAudioDevice(m_dev_id_in, 1);
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m_running = false;
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return true;
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}
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bool audio_async::clear() {
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if (!m_dev_id_in) {
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fprintf(stderr, "%s: no audio device to clear!\n", __func__);
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return false;
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}
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if (!m_running) {
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fprintf(stderr, "%s: not running!\n", __func__);
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return false;
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}
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{
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std::lock_guard<std::mutex> lock(m_mutex);
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m_audio_pos = 0;
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m_audio_len = 0;
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}
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return true;
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}
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// callback to be called by SDL
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void audio_async::callback(uint8_t * stream, int len) {
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if (!m_running) {
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return;
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}
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const size_t n_samples = len / sizeof(float);
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m_audio_new.resize(n_samples);
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memcpy(m_audio_new.data(), stream, n_samples * sizeof(float));
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//fprintf(stderr, "%s: %zu samples, pos %zu, len %zu\n", __func__, n_samples, m_audio_pos, m_audio_len);
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{
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std::lock_guard<std::mutex> lock(m_mutex);
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if (m_audio_pos + n_samples > m_audio.size()) {
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const size_t n0 = m_audio.size() - m_audio_pos;
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memcpy(&m_audio[m_audio_pos], stream, n0 * sizeof(float));
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memcpy(&m_audio[0], &stream[n0], (n_samples - n0) * sizeof(float));
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m_audio_pos = (m_audio_pos + n_samples) % m_audio.size();
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m_audio_len = m_audio.size();
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} else {
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memcpy(&m_audio[m_audio_pos], stream, n_samples * sizeof(float));
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m_audio_pos = (m_audio_pos + n_samples) % m_audio.size();
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m_audio_len = std::min(m_audio_len + n_samples, m_audio.size());
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}
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}
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}
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void audio_async::get(int ms, std::vector<float> & result) {
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if (!m_dev_id_in) {
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fprintf(stderr, "%s: no audio device to get audio from!\n", __func__);
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return;
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}
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if (!m_running) {
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fprintf(stderr, "%s: not running!\n", __func__);
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return;
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}
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result.clear();
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{
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std::lock_guard<std::mutex> lock(m_mutex);
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if (ms <= 0) {
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ms = m_len_ms;
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}
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size_t n_samples = (m_sample_rate * ms) / 1000;
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if (n_samples > m_audio_len) {
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n_samples = m_audio_len;
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}
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result.resize(n_samples);
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int s0 = m_audio_pos - n_samples;
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if (s0 < 0) {
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s0 += m_audio.size();
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}
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if (s0 + n_samples > m_audio.size()) {
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const size_t n0 = m_audio.size() - s0;
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memcpy(result.data(), &m_audio[s0], n0 * sizeof(float));
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memcpy(&result[n0], &m_audio[0], (n_samples - n0) * sizeof(float));
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} else {
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memcpy(result.data(), &m_audio[s0], n_samples * sizeof(float));
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}
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}
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}
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///////////////////////////
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void high_pass_filter(std::vector<float> & data, float cutoff, float sample_rate) {
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const float rc = 1.0f / (2.0f * M_PI * cutoff);
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const float dt = 1.0f / sample_rate;
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const float alpha = dt / (rc + dt);
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float y = data[0];
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for (size_t i = 1; i < data.size(); i++) {
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y = alpha * (y + data[i] - data[i - 1]);
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data[i] = y;
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}
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}
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bool vad_simple(std::vector<float> & pcmf32, int sample_rate, int last_ms, float vad_thold, float freq_thold, bool verbose) {
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const int n_samples = pcmf32.size();
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const int n_samples_last = (sample_rate * last_ms) / 1000;
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if (n_samples_last >= n_samples) {
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// not enough samples - assume no speech
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return false;
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}
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if (freq_thold > 0.0f) {
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high_pass_filter(pcmf32, freq_thold, sample_rate);
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}
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float energy_all = 0.0f;
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float energy_last = 0.0f;
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for (size_t i = 0; i < n_samples; i++) {
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energy_all += fabsf(pcmf32[i]);
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if (i >= n_samples - n_samples_last) {
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energy_last += fabsf(pcmf32[i]);
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}
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}
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energy_all /= n_samples;
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energy_last /= n_samples_last;
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if (verbose) {
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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);
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}
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if (energy_last > vad_thold*energy_all) {
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return false;
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}
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return true;
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}
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int main(int argc, char ** argv) {
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whisper_params params;
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@ -173,33 +419,46 @@ int main(int argc, char ** argv) {
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return 1;
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}
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params.keep_ms = std::min(params.keep_ms, params.step_ms); // cannot be more than step_ms
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const int n_samples_step = (params.step_ms *1e-3)*WHISPER_SAMPLE_RATE;
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const int n_samples_len = (params.length_ms*1e-3)*WHISPER_SAMPLE_RATE;
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const int n_samples_keep = (params.keep_ms *1e-3)*WHISPER_SAMPLE_RATE;
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const int n_samples_30s = (30000 *1e-3)*WHISPER_SAMPLE_RATE;
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const int n_new_line = params.length_ms / params.step_ms - 1; // number of steps to print new line
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const bool use_vad = n_samples_step <= 0;
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params.no_timestamps = !use_vad;
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params.no_context = use_vad;
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params.max_tokens = 0;
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// init audio
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if (!audio_sdl_init(params.capture_id)) {
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fprintf(stderr, "%s: audio_sdl_init() failed!\n", __func__);
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audio_async audio(params.length_ms);
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if (!audio.init(params.capture_id, WHISPER_SAMPLE_RATE)) {
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fprintf(stderr, "%s: audio.init() failed!\n", __func__);
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return 1;
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}
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audio.resume();
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// whisper init
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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
|
||||
{
|
||||
@ -211,23 +470,28 @@ int main(int argc, char ** argv) {
|
||||
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;
|
||||
@ -242,6 +506,9 @@ int main(int argc, char ** argv) {
|
||||
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
|
||||
@ -268,35 +535,64 @@ int main(int argc, char ** argv) {
|
||||
}
|
||||
|
||||
// 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);
|
||||
|
||||
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;
|
||||
}
|
||||
|
||||
while (SDL_GetQueuedAudioSize(g_dev_id_in) < n_samples*sizeof(float)) {
|
||||
SDL_Delay(1);
|
||||
}
|
||||
|
||||
const int n_samples_new = SDL_GetQueuedAudioSize(g_dev_id_in)/sizeof(float);
|
||||
|
||||
// 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));
|
||||
|
||||
// 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];
|
||||
}
|
||||
|
||||
SDL_DequeueAudio(g_dev_id_in, pcmf32.data() + n_samples_take, n_samples_new*sizeof(float));
|
||||
|
||||
pcmf32_old = pcmf32;
|
||||
|
||||
// run the inference
|
||||
{
|
||||
whisper_full_params wparams = whisper_full_default_params(WHISPER_SAMPLING_GREEDY);
|
||||
@ -307,7 +603,7 @@ int main(int argc, char ** argv) {
|
||||
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;
|
||||
@ -325,12 +621,21 @@ int main(int argc, char ** argv) {
|
||||
|
||||
// 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) {
|
||||
@ -358,11 +663,16 @@ int main(int argc, char ** argv) {
|
||||
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
|
||||
@ -384,9 +694,7 @@ int main(int argc, char ** argv) {
|
||||
}
|
||||
}
|
||||
|
||||
if (g_dev_id_in >= 0) {
|
||||
SDL_CloseAudioDevice(g_dev_id_in);
|
||||
}
|
||||
audio.pause();
|
||||
|
||||
whisper_print_timings(ctx);
|
||||
whisper_free(ctx);
|
||||
|
||||
Loading…
Reference in New Issue
Block a user