mirror of
https://github.com/ggerganov/whisper.cpp.git
synced 2025-06-12 20:18:08 +00:00
examples : refactor in order to reuse code and reduce duplication (#482)
* examples : refactor common code into a library * examples : refactor common SDL code into a library * make : update Makefile to use common libs * common : fix MSVC M_PI .. * addon.node : link common lib
This commit is contained in:
Georgi Gerganov
GitHub
@ -5,6 +5,5 @@ if (WHISPER_SUPPORT_SDL2)
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include(DefaultTargetOptions)
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target_include_directories(${TARGET} PRIVATE ${SDL2_INCLUDE_DIRS})
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target_link_libraries(${TARGET} PRIVATE whisper ${SDL2_LIBRARIES} ${CMAKE_THREAD_LIBS_INIT})
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target_link_libraries(${TARGET} PRIVATE common common-sdl whisper ${CMAKE_THREAD_LIBS_INIT})
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endif ()
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@ -3,19 +3,16 @@
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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 "common.h"
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#include "common-sdl.h"
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#include "whisper.h"
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#include <SDL.h>
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#include <SDL_audio.h>
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#include <atomic>
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#include <cassert>
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#include <cstdio>
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#include <string>
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#include <thread>
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#include <vector>
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#include <fstream>
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#include <mutex>
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// 500 -> 00:05.000
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// 6000 -> 01:00.000
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@ -116,306 +113,6 @@ void whisper_print_usage(int /*argc*/, char ** argv, const whisper_params & para
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fprintf(stderr, "\n");
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}
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//
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// SDL Audio capture
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//
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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 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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std::atomic_bool m_running;
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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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m_running = false;
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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 false;
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}
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SDL_SetHintWithPriority(SDL_HINT_AUDIO_RESAMPLING_MODE, "medium", SDL_HINT_OVERRIDE);
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{
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int nDevices = SDL_GetNumAudioDevices(SDL_TRUE);
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fprintf(stderr, "%s: found %d capture devices:\n", __func__, nDevices);
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for (int i = 0; i < nDevices; i++) {
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fprintf(stderr, "%s: - Capture device #%d: '%s'\n", __func__, i, SDL_GetAudioDeviceName(i, SDL_TRUE));
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}
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}
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SDL_AudioSpec capture_spec_requested;
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SDL_AudioSpec capture_spec_obtained;
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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 = 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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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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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 (!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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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__, 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 (int 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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@ -426,10 +123,10 @@ int main(int argc, char ** argv) {
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params.keep_ms = std::min(params.keep_ms, params.step_ms);
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params.length_ms = std::max(params.length_ms, params.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_samples_step = (1e-3*params.step_ms )*WHISPER_SAMPLE_RATE;
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const int n_samples_len = (1e-3*params.length_ms)*WHISPER_SAMPLE_RATE;
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const int n_samples_keep = (1e-3*params.keep_ms )*WHISPER_SAMPLE_RATE;
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const int n_samples_30s = (1e-3*30000.0 )*WHISPER_SAMPLE_RATE;
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const bool use_vad = n_samples_step <= 0; // sliding window mode uses VAD
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@ -517,23 +214,7 @@ int main(int argc, char ** argv) {
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// main audio loop
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while (is_running) {
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// handle Ctrl + C
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{
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SDL_Event event;
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while (SDL_PollEvent(&event)) {
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switch (event.type) {
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case SDL_QUIT:
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{
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is_running = false;
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} break;
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default:
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break;
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}
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}
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if (!is_running) {
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break;
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}
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}
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is_running = sdl_poll_events();
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if (!is_running) {
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break;
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@ -556,7 +237,7 @@ int main(int argc, char ** argv) {
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break;
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}
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SDL_Delay(1);
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std::this_thread::sleep_for(std::chrono::milliseconds(1));
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}
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const int n_samples_new = pcmf32_new.size();
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@ -587,7 +268,7 @@ int main(int argc, char ** argv) {
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audio.get(2000, pcmf32_new);
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if (vad_simple(pcmf32_new, WHISPER_SAMPLE_RATE, 1000, params.vad_thold, params.freq_thold, false)) {
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if (::vad_simple(pcmf32_new, WHISPER_SAMPLE_RATE, 1000, params.vad_thold, params.freq_thold, false)) {
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audio.get(params.length_ms, pcmf32);
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} else {
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std::this_thread::sleep_for(std::chrono::milliseconds(100));
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