stream : add sliding window mode

examples/stream/stream.cpp

@@ -1,6 +1,7 @@
 // Real-time speech recognition of input from a microphone
 //
 // A very quick-n-dirty implementation serving mainly as a proof of concept.
+//
 
 #include "whisper.h"
 
@@ -33,15 +34,19 @@ 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 no_context    = true;
     bool print_special = false;
-    bool no_timestamps = true;
+    bool no_context    = true;
+    bool no_timestamps = false;
 
     std::string language  = "en";
     std::string model     = "models/ggml-base.en.bin";
@@ -61,13 +66,16 @@ bool whisper_params_parse(int argc, char ** argv, whisper_params & params) {
         else if (arg == "-t"   || arg == "--threads")       { params.n_threads     = std::stoi(argv[++i]); }
         else if (                 arg == "--step")          { params.step_ms       = std::stoi(argv[++i]); }
         else if (                 arg == "--length")        { params.length_ms     = std::stoi(argv[++i]); }
+        else if (                 arg == "--keep")          { params.keep_ms       = std::stoi(argv[++i]); }
         else if (arg == "-c"   || arg == "--capture")       { params.capture_id    = std::stoi(argv[++i]); }
         else if (arg == "-mt"  || arg == "--max-tokens")    { params.max_tokens    = std::stoi(argv[++i]); }
         else if (arg == "-ac"  || arg == "--audio-ctx")     { params.audio_ctx     = std::stoi(argv[++i]); }
+        else if (arg == "-vth" || arg == "--vad-thold")     { params.vad_thold     = std::stof(argv[++i]); }
+        else if (arg == "-fth" || arg == "--freq-thold")    { params.freq_thold    = std::stof(argv[++i]); }
         else if (arg == "-su"  || arg == "--speed-up")      { params.speed_up      = true; }
         else if (arg == "-tr"  || arg == "--translate")     { params.translate     = true; }
-        else if (arg == "-kc"  || arg == "--keep-context")  { params.no_context    = false; }
         else if (arg == "-ps"  || arg == "--print-special") { params.print_special = true; }
+        else if (arg == "-kc"  || arg == "--keep-context")  { params.no_context    = false; }
         else if (arg == "-l"   || arg == "--language")      { params.language      = argv[++i]; }
         else if (arg == "-m"   || arg == "--model")         { params.model         = argv[++i]; }
         else if (arg == "-f"   || arg == "--file")          { params.fname_out     = argv[++i]; }
@@ -90,13 +98,16 @@ void whisper_print_usage(int argc, char ** argv, const whisper_params & params)
     fprintf(stderr, "  -t N,     --threads N     [%-7d] number of threads to use during computation\n", params.n_threads);
     fprintf(stderr, "            --step N        [%-7d] audio step size in milliseconds\n",             params.step_ms);
     fprintf(stderr, "            --length N      [%-7d] audio length in milliseconds\n",                params.length_ms);
+    fprintf(stderr, "            --keep N        [%-7d] audio to keep from previous step in ms\n",      params.keep_ms);
     fprintf(stderr, "  -c ID,    --capture ID    [%-7d] capture device ID\n",                           params.capture_id);
     fprintf(stderr, "  -mt N,    --max-tokens N  [%-7d] maximum number of tokens per audio chunk\n",    params.max_tokens);
     fprintf(stderr, "  -ac N,    --audio-ctx N   [%-7d] audio context size (0 - all)\n",                params.audio_ctx);
+    fprintf(stderr, "  -vth N,   --vad-thold N   [%-7.2f] voice activity detection threshold\n",        params.vad_thold);
+    fprintf(stderr, "  -fth N,   --freq-thold N  [%-7.2f] high-pass frequency cutoff\n",                params.freq_thold);
     fprintf(stderr, "  -su,      --speed-up      [%-7s] speed up audio by x2 (reduced accuracy)\n",     params.speed_up ? "true" : "false");
     fprintf(stderr, "  -tr,      --translate     [%-7s] translate from source language to english\n",   params.translate ? "true" : "false");
-    fprintf(stderr, "  -kc,      --keep-context  [%-7s] keep context between audio chunks\n",           params.no_context ? "false" : "true");
     fprintf(stderr, "  -ps,      --print-special [%-7s] print special tokens\n",                        params.print_special ? "true" : "false");
+    fprintf(stderr, "  -kc,      --keep-context  [%-7s] keep context between audio chunks\n",           params.no_context ? "false" : "true");
     fprintf(stderr, "  -l LANG,  --language LANG [%-7s] spoken language\n",                             params.language.c_str());
     fprintf(stderr, "  -m FNAME, --model FNAME   [%-7s] model path\n",                                  params.model.c_str());
     fprintf(stderr, "  -f FNAME, --file FNAME    [%-7s] text output file name\n",                       params.fname_out.c_str());
@@ -107,19 +118,56 @@ void whisper_print_usage(int argc, char ** argv, const whisper_params & params)
 // SDL Audio capture
 //
 
-SDL_AudioDeviceID g_dev_id_in = 0;
+class audio_async {
+public:
+    audio_async(int len_ms);
+    ~audio_async();
 
-bool audio_sdl_init(const int capture_id) {
+    bool init(int capture_id, int sample_rate);
-    if (g_dev_id_in) {
+
-        fprintf(stderr, "%s: already initialized\n", __func__);
+    // start capturing audio via the provided SDL callback
-        return false;
+    // keep last len_ms seconds of audio in a circular buffer
+    bool resume();
+    bool pause();
+    bool clear();
+
+    // callback to be called by SDL
+    void callback(uint8_t * stream, int len);
+
+    // get audio data from the circular buffer
+    void get(int ms, std::vector<float> & audio);
+
+private:
+    SDL_AudioDeviceID m_dev_id_in = 0;
+
+    int m_len_ms = 0;
+    int m_sample_rate = 0;
+
+    bool       m_running = false;
+    std::mutex m_mutex;
+
+    std::vector<float> m_audio;
+    std::vector<float> m_audio_new;
+    size_t             m_audio_pos = 0;
+    size_t             m_audio_len = 0;
+};
+
+audio_async::audio_async(int len_ms) {
+    m_len_ms = len_ms;
+}
+
+audio_async::~audio_async() {
+    if (m_dev_id_in) {
+        SDL_CloseAudioDevice(m_dev_id_in);
     }
+}
 
+bool audio_async::init(int capture_id, int sample_rate) {
     SDL_LogSetPriority(SDL_LOG_CATEGORY_APPLICATION, SDL_LOG_PRIORITY_INFO);
 
     if (SDL_Init(SDL_INIT_AUDIO) < 0) {
         SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, "Couldn't initialize SDL: %s\n", SDL_GetError());
-        return (1);
+        return false;
     }
 
     SDL_SetHintWithPriority(SDL_HINT_AUDIO_RESAMPLING_MODE, "medium", SDL_HINT_OVERRIDE);
@@ -138,34 +186,232 @@ bool audio_sdl_init(const int capture_id) {
     SDL_zero(capture_spec_requested);
     SDL_zero(capture_spec_obtained);
 
-    capture_spec_requested.freq     = WHISPER_SAMPLE_RATE;
+    capture_spec_requested.freq     = sample_rate;
     capture_spec_requested.format   = AUDIO_F32;
     capture_spec_requested.channels = 1;
     capture_spec_requested.samples  = 1024;
+    capture_spec_requested.callback = [](void * userdata, uint8_t * stream, int len) {
+        audio_async * audio = (audio_async *) userdata;
+        audio->callback(stream, len);
+    };
+    capture_spec_requested.userdata = this;
 
     if (capture_id >= 0) {
         fprintf(stderr, "%s: attempt to open capture device %d : '%s' ...\n", __func__, capture_id, SDL_GetAudioDeviceName(capture_id, SDL_TRUE));
-        g_dev_id_in = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(capture_id, SDL_TRUE), SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
+        m_dev_id_in = SDL_OpenAudioDevice(SDL_GetAudioDeviceName(capture_id, SDL_TRUE), SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
     } else {
         fprintf(stderr, "%s: attempt to open default capture device ...\n", __func__);
-        g_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
+        m_dev_id_in = SDL_OpenAudioDevice(nullptr, SDL_TRUE, &capture_spec_requested, &capture_spec_obtained, 0);
     }
-    if (!g_dev_id_in) {
+
+    if (!m_dev_id_in) {
         fprintf(stderr, "%s: couldn't open an audio device for capture: %s!\n", __func__, SDL_GetError());
-        g_dev_id_in = 0;
+        m_dev_id_in = 0;
+
+        return false;
     } else {
-        fprintf(stderr, "%s: obtained spec for input device (SDL Id = %d):\n", __func__, g_dev_id_in);
+        fprintf(stderr, "%s: 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:     - format:            %d (required: %d)\n",    __func__, capture_spec_obtained.format,
-        fprintf(stderr, "%s:     - channels:          %d (required: %d)\n", __func__, capture_spec_obtained.channels, capture_spec_requested.channels);
+                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;
 
@@ -173,33 +419,46 @@ int main(int argc, char ** argv) {
         return 1;
     }
 
+    params.keep_ms = std::min(params.keep_ms, params.step_ms); // cannot be more than step_ms
+
+    const int n_samples_step = (params.step_ms  *1e-3)*WHISPER_SAMPLE_RATE;
+    const int n_samples_len  = (params.length_ms*1e-3)*WHISPER_SAMPLE_RATE;
+    const int n_samples_keep = (params.keep_ms  *1e-3)*WHISPER_SAMPLE_RATE;
+    const int n_samples_30s  = (30000           *1e-3)*WHISPER_SAMPLE_RATE;
+
+    const int n_new_line = params.length_ms / params.step_ms - 1; // number of steps to print new line
+
+    const bool use_vad = n_samples_step <= 0;
+
+    params.no_timestamps = !use_vad;
+    params.no_context    = use_vad;
+    params.max_tokens    = 0;
+
     // init audio
 
-    if (!audio_sdl_init(params.capture_id)) {
+    audio_async audio(params.length_ms);
-        fprintf(stderr, "%s: audio_sdl_init() failed!\n", __func__);
+    if (!audio.init(params.capture_id, WHISPER_SAMPLE_RATE)) {
+        fprintf(stderr, "%s: audio.init() failed!\n", __func__);
         return 1;
     }
 
+    audio.resume();
+
+    // whisper init
+
     if (whisper_lang_id(params.language.c_str()) == -1) {
         fprintf(stderr, "error: unknown language '%s'\n", params.language.c_str());
         whisper_print_usage(argc, argv, params);
         exit(0);
     }
 
-    // whisper init
-
     struct whisper_context * ctx = whisper_init(params.model.c_str());
 
-    const int n_samples = (params.step_ms/1000.0)*WHISPER_SAMPLE_RATE;
+    std::vector<float> pcmf32    (n_samples_30s, 0.0f);
-    const int n_samples_len = (params.length_ms/1000.0)*WHISPER_SAMPLE_RATE;
+    std::vector<float> pcmf32_old(n_samples_30s, 0.0f);
-    const int n_samples_30s = 30*WHISPER_SAMPLE_RATE;
+    std::vector<float> pcmf32_new(n_samples_30s, 0.0f);
-    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<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,
+                n_samples_step,
-                float(n_samples)/WHISPER_SAMPLE_RATE,
+                float(n_samples_step)/WHISPER_SAMPLE_RATE,
-                float(n_samples_len)/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);
-        fprintf(stderr, "\n");
+        } else {
+            fprintf(stderr, "%s: using VAD, will transcribe on speech activity\n", __func__);
         }
 
-    SDL_PauseAudioDevice(g_dev_id_in, 0);
+        fprintf(stderr, "\n");
+    }
 
     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
@@ -268,19 +535,26 @@ int main(int argc, char ** argv) {
         }
 
         // process new audio
-        if (n_iter > 0 && SDL_GetQueuedAudioSize(g_dev_id_in) > 2*n_samples*sizeof(float)) {
+
+        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__);
-            SDL_ClearQueuedAudio(g_dev_id_in);
+                    audio.clear();
+                    continue;
                 }
 
-        while (SDL_GetQueuedAudioSize(g_dev_id_in) < n_samples*sizeof(float)) {
+                if ((int) pcmf32_new.size() >= n_samples_step) {
-            SDL_Delay(1);
+                    audio.clear();
+                    break;
                 }
 
-        const int n_samples_new = SDL_GetQueuedAudioSize(g_dev_id_in)/sizeof(float);
+                SDL_Delay(1);
+            }
 
-        // take one second from previous iteration
+            const int n_samples_new = pcmf32_new.size();
-        //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));
@@ -293,9 +567,31 @@ int main(int argc, char ** argv) {
                 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));
+            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
         {
@@ -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;
             {
+                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 = %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) {
+    audio.pause();
-        SDL_CloseAudioDevice(g_dev_id_in);
-    }
 
     whisper_print_timings(ctx);
     whisper_free(ctx);