Alibaba Cloud Model Studio:Qwen-Omni-Realtime

DashScope Python SDK

• Persiapkan lingkungan runtime

Versi Python Anda harus 3.10 atau lebih baru.

Pertama, instal pyaudio berdasarkan sistem operasi Anda.

brew install portaudio && pip install pyaudio

sudo yum install -y portaudio portaudio-devel && pip install pyaudio

pip install pyaudio

Setelah instalasi, instal dependensi menggunakan pip:

pip install websocket-client dashscope

• Pilih mode interaksi

  • Mode VAD (Deteksi aktivitas suara, deteksi otomatis awal/akhir ucapan)

    Server secara otomatis mendeteksi kapan pengguna mulai dan berhenti berbicara serta memberikan respons.

  • Mode manual (press-to-talk, release-to-send)

    Client mengontrol waktu bicara. Setelah pengguna selesai berbicara, client mengirim pesan ke server.

  Mode VAD

  Buat file Python baru bernama vad_dash.py dan salin kode berikut ke dalamnya:

  vad_dash.py

  # Dependensi: dashscope >= 1.23.9, pyaudio
  import os
  import base64
  import time
  import pyaudio
  from dashscope.audio.qwen_omni import MultiModality, AudioFormat,OmniRealtimeCallback,OmniRealtimeConversation
  import dashscope
  
  # Parameter konfigurasi: endpoint, API key, voice, model, peran model
  # Tentukan wilayah. Atur ke 'intl' untuk Internasional (Singapura) atau 'cn' untuk China (Beijing).
  region = 'intl'
  base_domain = 'dashscope-intl.aliyuncs.com' if region == 'intl' else 'dashscope.aliyuncs.com'
  url = f'wss://{base_domain}/api-ws/v1/realtime'
  # Konfigurasikan API key. Jika Anda belum menyetel variabel lingkungan, ganti baris berikut dengan dashscope.api_key = "sk-xxx"
  dashscope.api_key = os.getenv('DASHSCOPE_API_KEY')
  # Tentukan suara
  voice = 'Cherry'
  # Tentukan model
  model = 'qwen3.5-omni-plus-realtime'
  # Tentukan peran model
  instructions = "You are Xiaoyun, a personal assistant. Please answer the user's questions in a humorous and witty way."
  class SimpleCallback(OmniRealtimeCallback):
      def __init__(self, pya):
          self.pya = pya
          self.out = None
      def on_open(self):
          # Inisialisasi aliran output audio
          self.out = self.pya.open(
              format=pyaudio.paInt16,
              channels=1,
              rate=24000,
              output=True
          )
      def on_event(self, response):
          if response['type'] == 'response.audio.delta':
              # Putar audio
              self.out.write(base64.b64decode(response['delta']))
          elif response['type'] == 'conversation.item.input_audio_transcription.completed':
              # Cetak teks hasil transkripsi
              print(f"[User] {response['transcript']}")
          elif response['type'] == 'response.audio_transcript.done':
              # Cetak teks balasan asisten
              print(f"[LLM] {response['transcript']}")
  
  # 1. Inisialisasi perangkat audio
  pya = pyaudio.PyAudio()
  # 2. Buat fungsi callback dan sesi
  callback = SimpleCallback(pya)
  conv = OmniRealtimeConversation(model=model, callback=callback, url=url)
  # 3. Bangun koneksi dan konfigurasikan sesi
  conv.connect()
  conv.update_session(output_modalities=[MultiModality.AUDIO, MultiModality.TEXT], voice=voice, instructions=instructions)
  # 4. Inisialisasi aliran input audio
  mic = pya.open(format=pyaudio.paInt16, channels=1, rate=16000, input=True)
  # 5. Loop utama untuk memproses input audio
  print("Conversation started. Speak into the microphone (Ctrl+C to exit)...")
  try:
      while True:
          audio_data = mic.read(3200, exception_on_overflow=False)
          conv.append_audio(base64.b64encode(audio_data).decode())
          time.sleep(0.01)
  except KeyboardInterrupt:
      # Bersihkan resource
      conv.close()
      mic.close()
      callback.out.close()
      pya.terminate()
      print("\nConversation ended")

  Jalankan vad_dash.py untuk melakukan percakapan real-time dengan Qwen-Omni-Realtime melalui mikrofon Anda. Sistem mendeteksi awal/akhir ucapan dan secara otomatis mengirim data ke server tanpa intervensi manual.

  Mode manual

  Buat file Python baru bernama manual_dash.py dan salin kode berikut ke dalamnya:

  manual_dash.py

  # Dependensi: dashscope >= 1.23.9, pyaudio.
  import os
  import base64
  import sys
  import threading
  import pyaudio
  from dashscope.audio.qwen_omni import *
  import dashscope
  
  # Jika Anda belum menyetel variabel lingkungan, ganti baris berikut dengan API key Anda: dashscope.api_key = "sk-xxx"
  dashscope.api_key = os.getenv('DASHSCOPE_API_KEY')
  voice = 'Cherry'
  
  class MyCallback(OmniRealtimeCallback):
      """Callback minimal: Menginisialisasi speaker setelah koneksi dan langsung memutar audio yang dikembalikan."""
      def __init__(self, ctx):
          super().__init__()
          self.ctx = ctx
  
      def on_open(self) -> None:
          # Inisialisasi PyAudio dan speaker (24k/mono/16bit) setelah koneksi.
          print('connection opened')
          try:
              self.ctx['pya'] = pyaudio.PyAudio()
              self.ctx['out'] = self.ctx['pya'].open(
                  format=pyaudio.paInt16,
                  channels=1,
                  rate=24000,
                  output=True
              )
              print('audio output initialized')
          except Exception as e:
              print('[Error] audio init failed: {}'.format(e))
  
      def on_close(self, close_status_code, close_msg) -> None:
          print('connection closed with code: {}, msg: {}'.format(close_status_code, close_msg))
          sys.exit(0)
  
      def on_event(self, response: str) -> None:
          try:
              t = response['type']
              handlers = {
                  'session.created': lambda r: print('start session: {}'.format(r['session']['id'])),
                  'conversation.item.input_audio_transcription.completed': lambda r: print('question: {}'.format(r['transcript'])),
                  'response.audio_transcript.delta': lambda r: print('llm text: {}'.format(r['delta'])),
                  'response.audio.delta': self._play_audio,
                  'response.done': self._response_done,
              }
              h = handlers.get(t)
              if h:
                  h(response)
          except Exception as e:
              print('[Error] {}'.format(e))
  
      def _play_audio(self, response):
          # Decode Base64 dan tulis ke aliran output untuk diputar.
          if self.ctx['out'] is None:
              return
          try:
              data = base64.b64decode(response['delta'])
              self.ctx['out'].write(data)
          except Exception as e:
              print('[Error] audio playback failed: {}'.format(e))
  
      def _response_done(self, response):
          # Tandai putaran saat ini selesai agar loop utama menunggu.
          if self.ctx['conv'] is not None:
              print('[Metric] response: {}, first text delay: {}, first audio delay: {}'.format(
                  self.ctx['conv'].get_last_response_id(),
                  self.ctx['conv'].get_last_first_text_delay(),
                  self.ctx['conv'].get_last_first_audio_delay(),
              ))
          if self.ctx['resp_done'] is not None:
              self.ctx['resp_done'].set()
  
  def shutdown_ctx(ctx):
      """Lepaskan resource audio dan PyAudio secara aman."""
      try:
          if ctx['out'] is not None:
              ctx['out'].close()
              ctx['out'] = None
      except Exception:
          pass
      try:
          if ctx['pya'] is not None:
              ctx['pya'].terminate()
              ctx['pya'] = None
      except Exception:
          pass
  
  
  def record_until_enter(pya_inst: pyaudio.PyAudio, sample_rate=16000, chunk_size=3200):
      """Tekan Enter untuk menghentikan perekaman dan mengembalikan byte PCM."""
      frames = []
      stop_evt = threading.Event()
  
      stream = pya_inst.open(
          format=pyaudio.paInt16,
          channels=1,
          rate=sample_rate,
          input=True,
          frames_per_buffer=chunk_size
      )
  
      def _reader():
          while not stop_evt.is_set():
              try:
                  frames.append(stream.read(chunk_size, exception_on_overflow=False))
              except Exception:
                  break
  
      t = threading.Thread(target=_reader, daemon=True)
      t.start()
      input()  # Pengguna menekan Enter lagi untuk menghentikan perekaman.
      stop_evt.set()
      t.join(timeout=1.0)
      try:
          stream.close()
      except Exception:
          pass
      return b''.join(frames)
  
  
  if __name__  == '__main__':
      print('Initializing ...')
      # Konteks runtime: Menyimpan handle audio dan sesi.
      ctx = {'pya': None, 'out': None, 'conv': None, 'resp_done': threading.Event()}
      callback = MyCallback(ctx)
      conversation = OmniRealtimeConversation(
          model='qwen3.5-omni-plus-realtime',
          callback=callback,
          # Berikut adalah URL untuk wilayah Internasional (Singapura). Jika Anda menggunakan model di China (Beijing), ganti URL dengan wss://dashscope.aliyuncs.com/api-ws/v1/realtime
          url="wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime",
      )
      try:
          conversation.connect()
      except Exception as e:
          print('[Error] connect failed: {}'.format(e))
          sys.exit(1)
  
      ctx['conv'] = conversation
      # Konfigurasi sesi: Aktifkan output teks dan audio (nonaktifkan VAD sisi server, beralih ke perekaman manual).
      conversation.update_session(
          output_modalities=[MultiModality.AUDIO, MultiModality.TEXT],
          voice=voice,
          enable_input_audio_transcription=True,
          # Model untuk mentranskripsi audio input. Hanya gummy-realtime-v1 yang didukung.
          input_audio_transcription_model='gummy-realtime-v1',
          enable_turn_detection=False,
          instructions="You are Xiaoyun, a personal assistant. Please answer the user's questions accurately and friendly, always responding with a helpful attitude."
      )
  
      try:
          turn = 1
          while True:
              print(f"\n--- Turn {turn} ---")
              print("Press Enter to start recording (enter q to exit)...")
              user_input = input()
              if user_input.strip().lower() in ['q', 'quit']:
                  print("User requested to exit...")
                  break
              print("Recording... Press Enter again to stop.")
              if ctx['pya'] is None:
                  ctx['pya'] = pyaudio.PyAudio()
              recorded = record_until_enter(ctx['pya'])
              if not recorded:
                  print("No valid audio was recorded. Please try again.")
                  continue
              print(f"Successfully recorded audio: {len(recorded)} bytes. Sending...")
  
              # Kirim dalam potongan 3200-byte (sesuai dengan 16k/16bit/100ms).
              chunk_size = 3200
              for i in range(0, len(recorded), chunk_size):
                  chunk = recorded[i:i+chunk_size]
                  conversation.append_audio(base64.b64encode(chunk).decode('ascii'))
  
              print("Sending complete. Waiting for model response...")
              ctx['resp_done'].clear()
              conversation.commit()
              conversation.create_response()
              ctx['resp_done'].wait()
              print('Audio playback complete.')
              turn += 1
      except KeyboardInterrupt:
          print("\nProgram interrupted by user.")
      finally:
          shutdown_ctx(ctx)
          print("Program exited.")

  Jalankan manual_dash.py. Tekan Enter untuk berbicara, lalu tekan Enter lagi untuk menerima respons audio model.

DashScope Java SDK

Pilih mode interaksi

• Mode VAD (Deteksi aktivitas suara, deteksi otomatis awal/akhir ucapan)

  API Realtime secara otomatis mendeteksi waktu bicara dan memberikan respons.

• Mode manual (press-to-talk, release-to-send)

  Client mengontrol waktu bicara. Setelah pengguna selesai berbicara, client mengirim pesan ke server.

import com.alibaba.dashscope.audio.omni.*;
import com.alibaba.dashscope.exception.NoApiKeyException;
import com.google.gson.JsonObject;
import javax.sound.sampled.*;
import java.nio.ByteBuffer;
import java.util.Arrays;
import java.util.Base64;
import java.util.Map;
import java.util.Queue;
import java.util.concurrent.ConcurrentLinkedQueue;
import java.util.concurrent.atomic.AtomicBoolean;

public class OmniServerVad {
    static class SequentialAudioPlayer {
        private final SourceDataLine line;
        private final Queue<byte[]> audioQueue = new ConcurrentLinkedQueue<>();
        private final Thread playerThread;
        private final AtomicBoolean shouldStop = new AtomicBoolean(false);

        public SequentialAudioPlayer() throws LineUnavailableException {
            AudioFormat format = new AudioFormat(24000, 16, 1, true, false);
            line = AudioSystem.getSourceDataLine(format);
            line.open(format);
            line.start();

            playerThread = new Thread(() -> {
                while (!shouldStop.get()) {
                    byte[] audio = audioQueue.poll();
                    if (audio != null) {
                        line.write(audio, 0, audio.length);
                    } else {
                        try { Thread.sleep(10); } catch (InterruptedException ignored) {}
                    }
                }
            }, "AudioPlayer");
            playerThread.start();
        }

        public void play(String base64Audio) {
            try {
                byte[] audio = Base64.getDecoder().decode(base64Audio);
                audioQueue.add(audio);
            } catch (Exception e) {
                System.err.println("Audio decoding failed: " + e.getMessage());
            }
        }

        public void cancel() {
            audioQueue.clear();
            line.flush();
        }

        public void close() {
            shouldStop.set(true);
            try { playerThread.join(1000); } catch (InterruptedException ignored) {}
            line.drain();
            line.close();
        }
    }

    public static void main(String[] args) {
        try {
            SequentialAudioPlayer player = new SequentialAudioPlayer();
            AtomicBoolean userIsSpeaking = new AtomicBoolean(false);
            AtomicBoolean shouldStop = new AtomicBoolean(false);

            OmniRealtimeParam param = OmniRealtimeParam.builder()
                    .model("qwen3.5-omni-plus-realtime")
                    .apikey(System.getenv("DASHSCOPE_API_KEY"))
                    // Berikut adalah URL untuk wilayah Internasional (Singapura). Jika Anda menggunakan model di China (Beijing), ganti URL dengan wss://dashscope.aliyuncs.com/api-ws/v1/realtime
                    .url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
                    .build();

            OmniRealtimeConversation conversation = new OmniRealtimeConversation(param, new OmniRealtimeCallback() {
                @Override public void onOpen() {
                    System.out.println("Connection established");
                }
                @Override public void onClose(int code, String reason) {
                    System.out.println("Connection closed (" + code + "): " + reason);
                    shouldStop.set(true);
                }
                @Override public void onEvent(JsonObject event) {
                    handleEvent(event, player, userIsSpeaking);
                }
            });

            conversation.connect();
            conversation.updateSession(OmniRealtimeConfig.builder()
                    .modalities(Arrays.asList(OmniRealtimeModality.AUDIO, OmniRealtimeModality.TEXT))
                    .voice("Cherry")
                    .enableTurnDetection(true)
                    .enableInputAudioTranscription(true)
                    .parameters(Map.of("instructions",
                            "You are an AI customer service agent for a five-star hotel. Answer customer inquiries about room types, facilities, prices, and booking policies accurately and friendly. Always respond with a professional and helpful attitude. Do not provide unconfirmed information or information beyond the scope of the hotel's services."))
                    .build()
            );

            System.out.println("Please start speaking (automatic detection of speech start/end, press Ctrl+C to exit)...");
            AudioFormat format = new AudioFormat(16000, 16, 1, true, false);
            TargetDataLine mic = AudioSystem.getTargetDataLine(format);
            mic.open(format);
            mic.start();

            ByteBuffer buffer = ByteBuffer.allocate(3200);
            while (!shouldStop.get()) {
                int bytesRead = mic.read(buffer.array(), 0, buffer.capacity());
                if (bytesRead > 0) {
                    try {
                        conversation.appendAudio(Base64.getEncoder().encodeToString(buffer.array()));
                    } catch (Exception e) {
                        if (e.getMessage() != null && e.getMessage().contains("closed")) {
                            System.out.println("Conversation closed. Stopping recording.");
                            break;
                        }
                    }
                }
                Thread.sleep(20);
            }

            conversation.close(1000, "Normal exit");
            player.close();
            mic.close();
            System.out.println("\nProgram exited.");

        } catch (NoApiKeyException e) {
            System.err.println("API KEY not found: Please set the DASHSCOPE_API_KEY environment variable.");
            System.exit(1);
        } catch (Exception e) {
            e.printStackTrace();
        }
    }

    private static void handleEvent(JsonObject event, SequentialAudioPlayer player, AtomicBoolean userIsSpeaking) {
        String type = event.get("type").getAsString();
        switch (type) {
            case "input_audio_buffer.speech_started":
                System.out.println("\n[User started speaking]");
                player.cancel();
                userIsSpeaking.set(true);
                break;
            case "input_audio_buffer.speech_stopped":
                System.out.println("[User stopped speaking]");
                userIsSpeaking.set(false);
                break;
            case "response.audio.delta":
                if (!userIsSpeaking.get()) {
                    player.play(event.get("delta").getAsString());
                }
                break;
            case "conversation.item.input_audio_transcription.completed":
                System.out.println("User: " + event.get("transcript").getAsString());
                break;
            case "response.audio_transcript.delta":
                System.out.print(event.get("delta").getAsString());
                break;
            case "response.done":
                System.out.println("Response complete");
                break;
        }
    }
}

// DashScope Java SDK versi 2.20.9 atau lebih baru

import com.alibaba.dashscope.audio.omni.*;
import com.alibaba.dashscope.exception.NoApiKeyException;
import com.google.gson.JsonObject;
import javax.sound.sampled.*;
import java.io.ByteArrayOutputStream;
import java.io.IOException;
import java.util.Arrays;
import java.util.Base64;
import java.util.HashMap;
import java.util.Queue;
import java.util.concurrent.ConcurrentLinkedQueue;
import java.util.concurrent.CountDownLatch;
import java.util.concurrent.atomic.AtomicBoolean;
import java.util.concurrent.atomic.AtomicReference;

public class Main {
    // Definisi kelas RealtimePcmPlayer dimulai
    public static class RealtimePcmPlayer {
        private int sampleRate;
        private SourceDataLine line;
        private AudioFormat audioFormat;
        private Thread decoderThread;
        private Thread playerThread;
        private AtomicBoolean stopped = new AtomicBoolean(false);
        private Queue<String> b64AudioBuffer = new ConcurrentLinkedQueue<>();
        private Queue<byte[]> RawAudioBuffer = new ConcurrentLinkedQueue<>();

        // Konstruktor menginisialisasi format audio dan jalur audio.
        public RealtimePcmPlayer(int sampleRate) throws LineUnavailableException {
            this.sampleRate = sampleRate;
            this.audioFormat = new AudioFormat(this.sampleRate, 16, 1, true, false);
            DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
            line = (SourceDataLine) AudioSystem.getLine(info);
            line.open(audioFormat);
            line.start();
            decoderThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        String b64Audio = b64AudioBuffer.poll();
                        if (b64Audio != null) {
                            byte[] rawAudio = Base64.getDecoder().decode(b64Audio);
                            RawAudioBuffer.add(rawAudio);
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            playerThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        byte[] rawAudio = RawAudioBuffer.poll();
                        if (rawAudio != null) {
                            try {
                                playChunk(rawAudio);
                            } catch (IOException e) {
                                throw new RuntimeException(e);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            decoderThread.start();
            playerThread.start();
        }

        // Memutar potongan audio dan memblokir hingga pemutaran selesai.
        private void playChunk(byte[] chunk) throws IOException, InterruptedException {
            if (chunk == null || chunk.length == 0) return;

            int bytesWritten = 0;
            while (bytesWritten < chunk.length) {
                bytesWritten += line.write(chunk, bytesWritten, chunk.length - bytesWritten);
            }
            int audioLength = chunk.length / (this.sampleRate*2/1000);
            // Tunggu hingga audio dalam buffer selesai diputar.
            Thread.sleep(audioLength - 10);
        }

        public void write(String b64Audio) {
            b64AudioBuffer.add(b64Audio);
        }

        public void cancel() {
            b64AudioBuffer.clear();
            RawAudioBuffer.clear();
        }

        public void waitForComplete() throws InterruptedException {
            while (!b64AudioBuffer.isEmpty() || !RawAudioBuffer.isEmpty()) {
                Thread.sleep(100);
            }
            line.drain();
        }

        public void shutdown() throws InterruptedException {
            stopped.set(true);
            decoderThread.join();
            playerThread.join();
            if (line != null && line.isRunning()) {
                line.drain();
                line.close();
            }
        }
    } // Definisi kelas RealtimePcmPlayer berakhir
    // Tambahkan metode perekaman
    private static void recordAndSend(TargetDataLine line, OmniRealtimeConversation conversation) throws IOException {
        ByteArrayOutputStream out = new ByteArrayOutputStream();
        byte[] buffer = new byte[3200];
        AtomicBoolean stopRecording = new AtomicBoolean(false);

        // Mulai thread untuk mendengarkan tombol Enter.
        Thread enterKeyListener = new Thread(() -> {
            try {
                System.in.read();
                stopRecording.set(true);
            } catch (IOException e) {
                e.printStackTrace();
            }
        });
        enterKeyListener.start();

        // Loop perekaman
        while (!stopRecording.get()) {
            int count = line.read(buffer, 0, buffer.length);
            if (count > 0) {
                out.write(buffer, 0, count);
            }
        }

        // Kirim data yang direkam.
        byte[] audioData = out.toByteArray();
        String audioB64 = Base64.getEncoder().encodeToString(audioData);
        conversation.appendAudio(audioB64);
        out.close();
    }

    public static void main(String[] args) throws InterruptedException, LineUnavailableException {
        OmniRealtimeParam param = OmniRealtimeParam.builder()
                .model("qwen3.5-omni-plus-realtime")
                // API key untuk Singapura dan Beijing berbeda. Untuk mendapatkan API key, lihat https://www.alibabacloud.com/help/en/model-studio/get-api-key
                // Jika Anda belum mengonfigurasi variabel lingkungan, ganti baris berikut dengan API key Model Studio Anda: .apikey("sk-xxx")
                .apikey(System.getenv("DASHSCOPE_API_KEY"))
                //Berikut adalah URL untuk wilayah Singapura. Jika Anda menggunakan model di wilayah Beijing, ganti URL dengan wss://dashscope.aliyuncs.com/api-ws/v1/realtime
                .url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
                .build();
        AtomicReference<CountDownLatch> responseDoneLatch = new AtomicReference<>(null);
        responseDoneLatch.set(new CountDownLatch(1));

        RealtimePcmPlayer audioPlayer = new RealtimePcmPlayer(24000);
        final AtomicReference<OmniRealtimeConversation> conversationRef = new AtomicReference<>(null);
        OmniRealtimeConversation conversation = new OmniRealtimeConversation(param, new OmniRealtimeCallback() {
            @Override
            public void onOpen() {
                System.out.println("connection opened");
            }
            @Override
            public void onEvent(JsonObject message) {
                String type = message.get("type").getAsString();
                switch(type) {
                    case "session.created":
                        System.out.println("start session: " + message.get("session").getAsJsonObject().get("id").getAsString());
                        break;
                    case "conversation.item.input_audio_transcription.completed":
                        System.out.println("question: " + message.get("transcript").getAsString());
                        break;
                    case "response.audio_transcript.delta":
                        System.out.println("got llm response delta: " + message.get("delta").getAsString());
                        break;
                    case "response.audio.delta":
                        String recvAudioB64 = message.get("delta").getAsString();
                        audioPlayer.write(recvAudioB64);
                        break;
                    case "response.done":
                        System.out.println("======RESPONSE DONE======");
                        if (conversationRef.get() != null) {
                            System.out.println("[Metric] response: " + conversationRef.get().getResponseId() +
                                    ", first text delay: " + conversationRef.get().getFirstTextDelay() +
                                    " ms, first audio delay: " + conversationRef.get().getFirstAudioDelay() + " ms");
                        }
                        responseDoneLatch.get().countDown();
                        break;
                    default:
                        break;
                }
            }
            @Override
            public void onClose(int code, String reason) {
                System.out.println("connection closed code: " + code + ", reason: " + reason);
            }
        });
        conversationRef.set(conversation);
        try {
            conversation.connect();
        } catch (NoApiKeyException e) {
            throw new RuntimeException(e);
        }
        OmniRealtimeConfig config = OmniRealtimeConfig.builder()
                .modalities(Arrays.asList(OmniRealtimeModality.AUDIO, OmniRealtimeModality.TEXT))
                .voice("Cherry")
                .enableTurnDetection(false)
                // Tetapkan peran model.
                .parameters(new HashMap<String, Object>() {{
                    put("instructions","You are Xiaoyun, a personal assistant. Please answer the user's questions accurately and friendly, always responding with a helpful attitude.");
                }})
                .build();
        conversation.updateSession(config);

        // Tambahkan fungsionalitas perekaman mikrofon.
        AudioFormat format = new AudioFormat(16000, 16, 1, true, false);
        DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);

        if (!AudioSystem.isLineSupported(info)) {
            System.out.println("Line not supported");
            return;
        }

        TargetDataLine line = null;
        try {
            line = (TargetDataLine) AudioSystem.getLine(info);
            line.open(format);
            line.start();

            while (true) {
                System.out.println("Press Enter to start recording...");
                System.in.read();
                System.out.println("Recording started. Please speak... Press Enter again to stop recording and send.");
                recordAndSend(line, conversation);
                conversation.commit();
                conversation.createResponse(null, null);
                // Reset latch untuk menunggu berikutnya.
                responseDoneLatch.set(new CountDownLatch(1));
            }
        } catch (LineUnavailableException | IOException e) {
            e.printStackTrace();
        } finally {
            if (line != null) {
                line.stop();
                line.close();
            }
        }
    }}

WebSocket (Python)

• Persiapkan lingkungan runtime

  Versi Python Anda harus 3.10 atau lebih baru.

  Pertama, instal pyaudio berdasarkan sistem operasi Anda.

  macOS

  brew install portaudio && pip install pyaudio

  Debian/Ubuntu

  sudo apt-get install python3-pyaudio
  
  or
  
  pip install pyaudio

  Kami merekomendasikan menggunakan pip install pyaudio. Jika instalasi gagal, pertama instal dependensi portaudio untuk OS Anda.

  CentOS

  sudo yum install -y portaudio portaudio-devel && pip install pyaudio

  Windows

  pip install pyaudio

  Setelah instalasi, instal dependensi terkait WebSocket menggunakan pip:

  pip install websockets==15.0.1

• Buat client

  Buat file Python baru bernama omni_realtime_client.py di direktori lokal Anda dan salin kode berikut ke dalamnya:

  omni_realtime_client.py

  import asyncio
  import websockets
  import json
  import base64
  import time
  from typing import Optional, Callable, List, Dict, Any
  from enum import Enum
  
  class TurnDetectionMode(Enum):
      SERVER_VAD = "server_vad"
      MANUAL = "manual"
  
  class OmniRealtimeClient:
  
      def __init__(
              self,
              base_url,
              api_key: str,
              model: str = "",
              voice: str = "Ethan",
              instructions: str = "You are a helpful assistant.",
              turn_detection_mode: TurnDetectionMode = TurnDetectionMode.SERVER_VAD,
              on_text_delta: Optional[Callable[[str], None]] = None,
              on_audio_delta: Optional[Callable[[bytes], None]] = None,
              on_input_transcript: Optional[Callable[[str], None]] = None,
              on_output_transcript: Optional[Callable[[str], None]] = None,
              extra_event_handlers: Optional[Dict[str, Callable[[Dict[str, Any]], None]]] = None
      ):
          self.base_url = base_url
          self.api_key = api_key
          self.model = model
          self.voice = voice
          self.instructions = instructions
          self.ws = None
          self.on_text_delta = on_text_delta
          self.on_audio_delta = on_audio_delta
          self.on_input_transcript = on_input_transcript
          self.on_output_transcript = on_output_transcript
          self.turn_detection_mode = turn_detection_mode
          self.extra_event_handlers = extra_event_handlers or {}
  
          # Status respons saat ini
          self._current_response_id = None
          self._current_item_id = None
          self._is_responding = False
          # Status pencetakan transkrip input/output
          self._print_input_transcript = True
          self._output_transcript_buffer = ""
  
      async def connect(self) -> None:
          """Membangun koneksi WebSocket dengan API Realtime."""
          url = f"{self.base_url}?model={self.model}"
          headers = {
              "Authorization": f"Bearer {self.api_key}"
          }
          self.ws = await websockets.connect(url, additional_headers=headers)
  
          # Konfigurasi sesi
          session_config = {
              "modalities": ["text", "audio"],
              "voice": self.voice,
              "instructions": self.instructions,
              "input_audio_format": "pcm",
              "output_audio_format": "pcm",
              "input_audio_transcription": {
                  "model": "gummy-realtime-v1"
              }
          }
  
          if self.turn_detection_mode == TurnDetectionMode.MANUAL:
              session_config['turn_detection'] = None
              await self.update_session(session_config)
          elif self.turn_detection_mode == TurnDetectionMode.SERVER_VAD:
              session_config['turn_detection'] = {
                  "type": "server_vad",
                  "threshold": 0.1,
                  "prefix_padding_ms": 500,
                  "silence_duration_ms": 900
              }
              await self.update_session(session_config)
          else:
              raise ValueError(f"Invalid turn detection mode: {self.turn_detection_mode}")
  
      async def send_event(self, event) -> None:
          event['event_id'] = "event_" + str(int(time.time() * 1000))
          await self.ws.send(json.dumps(event))
  
      async def update_session(self, config: Dict[str, Any]) -> None:
          """Memperbarui konfigurasi sesi."""
          event = {
              "type": "session.update",
              "session": config
          }
          await self.send_event(event)
  
      async def stream_audio(self, audio_chunk: bytes) -> None:
          """Streaming data audio mentah ke API."""
          # Hanya PCM 16-bit, 16 kHz, mono yang didukung.
          audio_b64 = base64.b64encode(audio_chunk).decode()
          append_event = {
              "type": "input_audio_buffer.append",
              "audio": audio_b64
          }
          await self.send_event(append_event)
  
      async def commit_audio_buffer(self) -> None:
          """Commit buffer audio untuk memicu pemrosesan."""
          event = {
              "type": "input_audio_buffer.commit"
          }
          await self.send_event(event)
  
      async def append_image(self, image_chunk: bytes) -> None:
          """Menambahkan data gambar ke buffer gambar.
          Data gambar dapat berasal dari file lokal atau aliran video real-time.
          Catatan:
              - Format gambar harus JPG atau JPEG. Kami merekomendasikan resolusi 480p atau 720p. Resolusi maksimum yang didukung adalah 1080p.
              - Satu gambar tidak boleh melebihi 500 KB.
              - Encode data gambar ke Base64 sebelum mengirim.
              - Kami merekomendasikan mengirim gambar tidak lebih dari 1 frame per detik.
              - Anda harus mengirim data audio setidaknya sekali sebelum mengirim data gambar.
          """
          image_b64 = base64.b64encode(image_chunk).decode()
          event = {
              "type": "input_image_buffer.append",
              "image": image_b64
          }
          await self.send_event(event)
  
      async def create_response(self) -> None:
          """Meminta API untuk menghasilkan respons (hanya diperlukan dalam mode manual)."""
          event = {
              "type": "response.create"
          }
          await self.send_event(event)
  
      async def cancel_response(self) -> None:
          """Membatalkan respons saat ini."""
          event = {
              "type": "response.cancel"
          }
          await self.send_event(event)
  
      async def handle_interruption(self):
          """Menangani interupsi pengguna terhadap respons saat ini."""
          if not self._is_responding:
              return
          # 1. Batalkan respons saat ini.
          if self._current_response_id:
              await self.cancel_response()
  
          self._is_responding = False
          self._current_response_id = None
          self._current_item_id = None
  
      async def handle_messages(self) -> None:
          try:
              async for message in self.ws:
                  event = json.loads(message)
                  event_type = event.get("type")
                  if event_type == "error":
                      print(" Error: ", event['error'])
                      continue
                  elif event_type == "response.created":
                      self._current_response_id = event.get("response", {}).get("id")
                      self._is_responding = True
                  elif event_type == "response.output_item.added":
                      self._current_item_id = event.get("item", {}).get("id")
                  elif event_type == "response.done":
                      self._is_responding = False
                      self._current_response_id = None
                      self._current_item_id = None
                  elif event_type == "input_audio_buffer.speech_started":
                      print("Speech start detected")
                      if self._is_responding:
                          print("Handling interruption")
                          await self.handle_interruption()
                  elif event_type == "input_audio_buffer.speech_stopped":
                      print("Speech end detected")
                  elif event_type == "response.text.delta":
                      if self.on_text_delta:
                          self.on_text_delta(event["delta"])
                  elif event_type == "response.audio.delta":
                      if self.on_audio_delta:
                          audio_bytes = base64.b64decode(event["delta"])
                          self.on_audio_delta(audio_bytes)
                  elif event_type == "conversation.item.input_audio_transcription.completed":
                      transcript = event.get("transcript", "")
                      print(f"User: {transcript}")
                      if self.on_input_transcript:
                          await asyncio.to_thread(self.on_input_transcript, transcript)
                          self._print_input_transcript = True
                  elif event_type == "response.audio_transcript.delta":
                      if self.on_output_transcript:
                          delta = event.get("delta", "")
                          if not self._print_input_transcript:
                              self._output_transcript_buffer += delta
                          else:
                              if self._output_transcript_buffer:
                                  await asyncio.to_thread(self.on_output_transcript, self._output_transcript_buffer)
                                  self._output_transcript_buffer = ""
                              await asyncio.to_thread(self.on_output_transcript, delta)
                  elif event_type == "response.audio_transcript.done":
                      print(f"LLM: {event.get('transcript', '')}")
                      self._print_input_transcript = False
                  elif event_type in self.extra_event_handlers:
                      self.extra_event_handlers[event_type](event)
          except websockets.exceptions.ConnectionClosed:
              print(" Connection closed")
          except Exception as e:
              print(" Error in message handling: ", str(e))
      async def close(self) -> None:
          """Menutup koneksi WebSocket."""
          if self.ws:
              await self.ws.close()

• Pilih mode interaksi

  • Mode VAD (Deteksi aktivitas suara, deteksi otomatis awal/akhir ucapan)

    API Realtime secara otomatis mendeteksi waktu bicara dan memberikan respons.

  • Mode manual (press-to-talk, release-to-send)

    Client mengontrol waktu bicara. Setelah pengguna selesai berbicara, client mengirim pesan ke server.

  Mode VAD

  Di direktori yang sama dengan omni_realtime_client.py, buat file Python lain bernama vad_mode.py dan salin kode berikut ke dalamnya:

  vad_mode.py

  # -- coding: utf-8 --
  import os, asyncio, pyaudio, queue, threading
  from omni_realtime_client import OmniRealtimeClient, TurnDetectionMode
  
  # Kelas pemutar audio (menangani interupsi)
  class AudioPlayer:
      def __init__(self, pyaudio_instance, rate=24000):
          self.stream = pyaudio_instance.open(format=pyaudio.paInt16, channels=1, rate=rate, output=True)
          self.queue = queue.Queue()
          self.stop_evt = threading.Event()
          self.interrupt_evt = threading.Event()
          threading.Thread(target=self._run, daemon=True).start()
  
      def _run(self):
          while not self.stop_evt.is_set():
              try:
                  data = self.queue.get(timeout=0.5)
                  if data is None: break
                  if not self.interrupt_evt.is_set(): self.stream.write(data)
                  self.queue.task_done()
              except queue.Empty: continue
  
      def add_audio(self, data): self.queue.put(data)
      def handle_interrupt(self): self.interrupt_evt.set(); self.queue.queue.clear()
      def stop(self): self.stop_evt.set(); self.queue.put(None); self.stream.stop_stream(); self.stream.close()
  
  # Rekam dari mikrofon dan kirim
  async def record_and_send(client):
      p = pyaudio.PyAudio()
      stream = p.open(format=pyaudio.paInt16, channels=1, rate=16000, input=True, frames_per_buffer=3200)
      print("Recording started. Please speak...")
      try:
          while True:
              audio_data = stream.read(3200)
              await client.stream_audio(audio_data)
              await asyncio.sleep(0.02)
      finally:
          stream.stop_stream(); stream.close(); p.terminate()
  
  async def main():
      p = pyaudio.PyAudio()
      player = AudioPlayer(pyaudio_instance=p)
  
      client = OmniRealtimeClient(
          # Berikut adalah base_url untuk wilayah Internasional (Singapura). base_url untuk wilayah China (Beijing) adalah wss://dashscope.aliyuncs.com/api-ws/v1/realtime
          base_url="wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime",
          api_key=os.environ.get("DASHSCOPE_API_KEY"),
          model="qwen3.5-omni-plus-realtime",
          voice="Cherry",
          instructions="You are Xiaoyun, a witty and humorous assistant.",
          turn_detection_mode=TurnDetectionMode.SERVER_VAD,
          on_text_delta=lambda t: print(f"\nAssistant: {t}", end="", flush=True),
          on_audio_delta=player.add_audio,
      )
  
      await client.connect()
      print("Connection successful. Starting real-time conversation...")
  
      # Jalankan secara konkuren
      await asyncio.gather(client.handle_messages(), record_and_send(client))
  
  if __name__ == "__main__":
      try:
          asyncio.run(main())
      except KeyboardInterrupt:
          print("\nProgram exited.")

  Jalankan vad_mode.py untuk melakukan percakapan real-time dengan model Realtime melalui mikrofon Anda. Sistem mendeteksi awal/akhir ucapan dan secara otomatis mengirim data ke server tanpa intervensi manual.

  Mode manual

  Di direktori yang sama dengan omni_realtime_client.py, buat file Python lain bernama manual_mode.py dan salin kode berikut ke dalamnya:

  manual_mode.py

  # -- coding: utf-8 --
  import os
  import asyncio
  import time
  import threading
  import queue
  import pyaudio
  from omni_realtime_client import OmniRealtimeClient, TurnDetectionMode
  
  
  class AudioPlayer:
      """Kelas pemutar audio real-time"""
  
      def __init__(self, sample_rate=24000, channels=1, sample_width=2):
          self.sample_rate = sample_rate
          self.channels = channels
          self.sample_width = sample_width  # 2 byte untuk 16-bit
          self.audio_queue = queue.Queue()
          self.is_playing = False
          self.play_thread = None
          self.pyaudio_instance = None
          self.stream = None
          self._lock = threading.Lock()  # Tambahkan lock untuk akses sinkronisasi
          self._last_data_time = time.time()  # Catat waktu data terakhir diterima
          self._response_done = False  # Tambahkan flag untuk menandakan penyelesaian respons
          self._waiting_for_response = False  # Flag untuk menandakan jika sedang menunggu respons server
          # Catat waktu data terakhir ditulis ke aliran audio dan durasi potongan audio terbaru untuk deteksi akhir pemutaran yang lebih akurat
          self._last_play_time = time.time()
          self._last_chunk_duration = 0.0
  
      def start(self):
          """Mulai pemutar audio"""
          with self._lock:
              if self.is_playing:
                  return
  
              self.is_playing = True
  
              try:
                  self.pyaudio_instance = pyaudio.PyAudio()
  
                  # Buat aliran output audio
                  self.stream = self.pyaudio_instance.open(
                      format=pyaudio.paInt16,  # 16-bit
                      channels=self.channels,
                      rate=self.sample_rate,
                      output=True,
                      frames_per_buffer=1024
                  )
  
                  # Mulai thread pemutaran
                  self.play_thread = threading.Thread(target=self._play_audio)
                  self.play_thread.daemon = True
                  self.play_thread.start()
  
                  print("Audio player started")
              except Exception as e:
                  print(f"Failed to start audio player: {e}")
                  self._cleanup_resources()
                  raise
  
      def stop(self):
          """Hentikan pemutar audio"""
          with self._lock:
              if not self.is_playing:
                  return
  
              self.is_playing = False
  
          # Kosongkan antrian
          while not self.audio_queue.empty():
              try:
                  self.audio_queue.get_nowait()
              except queue.Empty:
                  break
  
          # Tunggu thread pemutaran selesai (tunggu di luar lock untuk menghindari deadlock)
          if self.play_thread and self.play_thread.is_alive():
              self.play_thread.join(timeout=2.0)
  
          # Dapatkan lock lagi untuk membersihkan resource
          with self._lock:
              self._cleanup_resources()
  
          print("Audio player stopped")
  
      def _cleanup_resources(self):
          """Bersihkan resource audio (harus dipanggil dalam lock)"""
          try:
              # Tutup aliran audio
              if self.stream:
                  if not self.stream.is_stopped():
                      self.stream.stop_stream()
                  self.stream.close()
                  self.stream = None
          except Exception as e:
              print(f"Error closing audio stream: {e}")
  
          try:
              if self.pyaudio_instance:
                  self.pyaudio_instance.terminate()
                  self.pyaudio_instance = None
          except Exception as e:
              print(f"Error terminating PyAudio: {e}")
  
      def add_audio_data(self, audio_data):
          """Tambahkan data audio ke antrian pemutaran"""
          if self.is_playing and audio_data:
              self.audio_queue.put(audio_data)
              with self._lock:
                  self._last_data_time = time.time()  # Perbarui waktu data terakhir diterima
                  self._waiting_for_response = False  # Data diterima, tidak menunggu lagi
  
      def stop_receiving_data(self):
          """Tandai bahwa tidak ada data audio baru yang akan diterima"""
          with self._lock:
              self._response_done = True
              self._waiting_for_response = False  # Respons berakhir, tidak menunggu lagi
  
      def prepare_for_next_turn(self):
          """Reset status pemutar untuk giliran percakapan berikutnya."""
          with self._lock:
              self._response_done = False
              self._last_data_time = time.time()
              self._last_play_time = time.time()
              self._last_chunk_duration = 0.0
              self._waiting_for_response = True  # Mulai menunggu respons berikutnya
  
          # Kosongkan data audio yang tersisa dari giliran sebelumnya
          while not self.audio_queue.empty():
              try:
                  self.audio_queue.get_nowait()
              except queue.Empty:
                  break
  
      def is_finished_playing(self):
          """Periksa apakah semua data audio telah diputar"""
          with self._lock:
              queue_size = self.audio_queue.qsize()
              time_since_last_data = time.time() - self._last_data_time
              time_since_last_play = time.time() - self._last_play_time
  
              # ---------------------- Deteksi akhir cerdas ----------------------
              # 1. Lebih disukai: Jika server telah menandai penyelesaian dan antrian pemutaran kosong.
              #    Tunggu potongan audio terbaru selesai diputar (durasi potongan + toleransi 0.1 detik).
              if self._response_done and queue_size == 0:
                  min_wait = max(self._last_chunk_duration + 0.1, 0.5)  # Tunggu minimal 0.5 detik
                  if time_since_last_play >= min_wait:
                      return True
  
              # 2. Cadangan: Jika tidak ada data baru yang diterima dalam waktu lama dan antrian pemutaran kosong.
              #    Logika ini berfungsi sebagai pengaman jika server tidak secara eksplisit mengirim `response.done`.
              if not self._waiting_for_response and queue_size == 0 and time_since_last_data > 1.0:
                  print("\n(No new audio received for a while, assuming playback is finished)")
                  return True
  
              return False
  
      def _play_audio(self):
          """Thread pekerja untuk memutar data audio"""
          while True:
              # Periksa apakah harus berhenti
              with self._lock:
                  if not self.is_playing:
                      break
                  stream_ref = self.stream  # Dapatkan referensi ke aliran
  
              try:
                  # Dapatkan data audio dari antrian, dengan timeout 0.1 detik
                  audio_data = self.audio_queue.get(timeout=0.1)
  
                  # Periksa status dan validitas aliran lagi
                  with self._lock:
                      if self.is_playing and stream_ref and not stream_ref.is_stopped():
                          try:
                              # Putar data audio
                              stream_ref.write(audio_data)
                              # Perbarui informasi pemutaran terbaru
                              self._last_play_time = time.time()
                              self._last_chunk_duration = len(audio_data) / (
                                          self.channels * self.sample_width) / self.sample_rate
                          except Exception as e:
                              print(f"Error writing to audio stream: {e}")
                              break
  
                  # Tandai blok data ini telah diproses
                  self.audio_queue.task_done()
  
              except queue.Empty:
                  # Lanjutkan menunggu jika antrian kosong
                  continue
              except Exception as e:
                  print(f"Error playing audio: {e}")
                  break
  
  
  class MicrophoneRecorder:
      """Perekam mikrofon real-time"""
  
      def __init__(self, sample_rate=16000, channels=1, chunk_size=3200):
          self.sample_rate = sample_rate
          self.channels = channels
          self.chunk_size = chunk_size
          self.pyaudio_instance = None
          self.stream = None
          self.frames = []
          self._is_recording = False
          self._record_thread = None
  
      def _recording_thread(self):
          """Thread pekerja perekaman"""
          # Terus-menerus membaca data dari aliran audio selama _is_recording bernilai True
          while self._is_recording:
              try:
                  # Gunakan exception_on_overflow=False untuk menghindari crash karena luapan buffer
                  data = self.stream.read(self.chunk_size, exception_on_overflow=False)
                  self.frames.append(data)
              except (IOError, OSError) as e:
                  # Membaca dari aliran mungkin menimbulkan error saat ditutup
                  print(f"Error reading from recording stream, it might be closed: {e}")
                  break
  
      def start(self):
          """Mulai perekaman"""
          if self._is_recording:
              print("Recording is already in progress.")
              return
  
          self.frames = []
          self._is_recording = True
  
          try:
              self.pyaudio_instance = pyaudio.PyAudio()
              self.stream = self.pyaudio_instance.open(
                  format=pyaudio.paInt16,
                  channels=self.channels,
                  rate=self.sample_rate,
                  input=True,
                  frames_per_buffer=self.chunk_size
              )
  
              self._record_thread = threading.Thread(target=self._recording_thread)
              self._record_thread.daemon = True
              self._record_thread.start()
              print("Microphone recording started...")
          except Exception as e:
              print(f"Failed to start microphone: {e}")
              self._is_recording = False
              self._cleanup()
              raise
  
      def stop(self):
          """Hentikan perekaman dan kembalikan data audio"""
          if not self._is_recording:
              return None
  
          self._is_recording = False
  
          # Tunggu thread perekaman keluar dengan aman
          if self._record_thread:
              self._record_thread.join(timeout=1.0)
  
          self._cleanup()
  
          print("Microphone recording stopped.")
          return b''.join(self.frames)
  
      def _cleanup(self):
          """Bersihkan resource PyAudio dengan aman"""
          if self.stream:
              try:
                  if self.stream.is_active():
                      self.stream.stop_stream()
                  self.stream.close()
              except Exception as e:
                  print(f"Error closing audio stream: {e}")
  
          if self.pyaudio_instance:
              try:
                  self.pyaudio_instance.terminate()
              except Exception as e:
                  print(f"Error terminating PyAudio instance: {e}")
  
          self.stream = None
          self.pyaudio_instance = None
  
  
  async def interactive_test():
      """
      Skrip uji interaktif: Memungkinkan percakapan multi-putaran, dengan audio dan gambar dikirim di setiap giliran.
      """
      # ------------------- 1. Inisialisasi dan koneksi (sekali saja) -------------------
      # API key untuk Singapura dan Beijing berbeda. Untuk mendapatkan API key, lihat https://www.alibabacloud.com/help/en/model-studio/get-api-key
      api_key = os.environ.get("DASHSCOPE_API_KEY")
      if not api_key:
          print("Please set the DASHSCOPE_API_KEY environment variable.")
          return
  
      print("--- Real-time Multimodal Audio/Video Chat Client ---")
      print("Initializing audio player and client...")
  
      audio_player = AudioPlayer()
      audio_player.start()
  
      def on_audio_received(audio_data):
          audio_player.add_audio_data(audio_data)
  
      def on_response_done(event):
          print("\n(Received response end marker)")
          audio_player.stop_receiving_data()
  
      realtime_client = OmniRealtimeClient(
          # Berikut adalah base_url untuk wilayah Singapura. Jika Anda menggunakan model di wilayah Beijing, ganti base_url dengan wss://dashscope.aliyuncs.com/api-ws/v1/realtime
          base_url="wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime",
          api_key=api_key,
          model="qwen3.5-omni-plus-realtime",
          voice="Ethan",
          instructions="You are Xiaoyun, a personal assistant. Please answer the user's questions accurately and friendly, always responding with a helpful attitude.", # Tetapkan peran model
          on_text_delta=lambda text: print(f"Assistant reply: {text}", end="", flush=True),
          on_audio_delta=on_audio_received,
          turn_detection_mode=TurnDetectionMode.MANUAL,
          extra_event_handlers={"response.done": on_response_done}
      )
  
      message_handler_task = None
      try:
          await realtime_client.connect()
          print("Connected to the server. Enter 'q' or 'quit' to exit at any time.")
          message_handler_task = asyncio.create_task(realtime_client.handle_messages())
          await asyncio.sleep(0.5)
  
          turn_counter = 1
          # ------------------- 2. Loop percakapan multi-putaran -------------------
          while True:
              print(f"\n--- Turn {turn_counter} ---")
              audio_player.prepare_for_next_turn()
  
              recorded_audio = None
              image_paths = []
  
              # --- Dapatkan input pengguna: Rekam dari mikrofon ---
              loop = asyncio.get_event_loop()
              recorder = MicrophoneRecorder(sample_rate=16000)  # Laju sampel 16k direkomendasikan untuk pengenalan ucapan
  
              print("Ready to record. Press Enter to start recording (or enter 'q' to exit)...")
              user_input = await loop.run_in_executor(None, input)
              if user_input.strip().lower() in ['q', 'quit']:
                  print("User requested to exit...")
                  return
  
              try:
                  recorder.start()
              except Exception:
                  print("Could not start recording. Please check your microphone permissions and device. Skipping this turn.")
                  continue
  
              print("Recording... Press Enter again to stop.")
              await loop.run_in_executor(None, input)
  
              recorded_audio = recorder.stop()
  
              if not recorded_audio or len(recorded_audio) == 0:
                  print("No valid audio was recorded. Please start this turn again.")
                  continue
  
              # --- Dapatkan input gambar (opsional) ---
              # Fitur input gambar di bawah ini dikomentari dan dinonaktifkan sementara. Untuk mengaktifkannya, hapus komentar kode di bawah.
              # print("\nEnter the absolute path of an [image file] on each line (optional). When finished, enter 's' or press Enter to send the request.")
              # while True:
              #     path = input("Image path: ").strip()
              #     if path.lower() == 's' or path == '':
              #         break
              #     if path.lower() in ['q', 'quit']:
              #         print("User requested to exit...")
              #         return
              #
              #     if not os.path.isabs(path):
              #         print("Error: Please enter an absolute path.")
              #         continue
              #     if not os.path.exists(path):
              #         print(f"Error: File not found -> {path}")
              #         continue
              #     image_paths.append(path)
              #     print(f"Image added: {os.path.basename(path)}")
  
              # --- 3. Kirim data dan dapatkan respons ---
              print("\n--- Input Confirmation ---")
              print(f"Audio to process: 1 (from microphone), Images: {len(image_paths)}")
              print("------------------")
  
              # 3.1 Kirim rekaman audio
              try:
                  print(f"Sending microphone recording ({len(recorded_audio)} bytes)")
                  await realtime_client.stream_audio(recorded_audio)
                  await asyncio.sleep(0.1)
              except Exception as e:
                  print(f"Failed to send microphone recording: {e}")
                  continue
  
              # 3.2 Kirim semua file gambar
              # Kode pengiriman gambar di bawah ini dikomentari dan dinonaktifkan.
              # for i, path in enumerate(image_paths):
              #     try:
              #         with open(path, "rb") as f:
              #             data = f.read()
              #         print(f"Sending image {i+1}: {os.path.basename(path)} ({len(data)} bytes)")
              #         await realtime_client.append_image(data)
              #         await asyncio.sleep(0.1)
              #     except Exception as e:
              #         print(f"Failed to send image {os.path.basename(path)}: {e}")
  
              # 3.3 Kirim dan tunggu respons
              print("Submitting all inputs, requesting server response...")
              await realtime_client.commit_audio_buffer()
              await realtime_client.create_response()
  
              print("Waiting for and playing server response audio...")
              start_time = time.time()
              max_wait_time = 60
              while not audio_player.is_finished_playing():
                  if time.time() - start_time > max_wait_time:
                      print(f"\nWait timed out ({max_wait_time} seconds). Moving to the next turn.")
                      break
                  await asyncio.sleep(0.2)
  
              print("\nAudio playback for this turn is complete!")
              turn_counter += 1
  
      except (asyncio.CancelledError, KeyboardInterrupt):
          print("\nProgram was interrupted.")
      except Exception as e:
          print(f"An unhandled error occurred: {e}")
      finally:
          # ------------------- 4. Bersihkan resource -------------------
          print("\nClosing connection and cleaning up resources...")
          if message_handler_task and not message_handler_task.done():
              message_handler_task.cancel()
  
          if 'realtime_client' in locals() and realtime_client.ws and not realtime_client.ws.close:
              await realtime_client.close()
              print("Connection closed.")
  
          audio_player.stop()
          print("Program exited.")
  
  
  if __name__ == "__main__":
      try:
          asyncio.run(interactive_test())
      except KeyboardInterrupt:
          print("\nProgram was forcibly exited by the user.")

  Jalankan manual_mode.py. Tekan Enter untuk berbicara, lalu tekan Enter lagi untuk menerima respons audio model.

Mode VAD

Atur session.turn_detection ke "server_vad" dalam event session.update untuk mengaktifkan mode VAD. Dalam mode ini, server secara otomatis mendeteksi awal/akhir ucapan dan memberikan respons. Cocok untuk panggilan suara.

Alur interaksinya sebagai berikut:

1. Server mendeteksi awal ucapan dan mengirim event input_audio_buffer.speech_started.

2. Client dapat mengirim event input_audio_buffer.append dan input_image_buffer.append kapan saja untuk menambahkan audio dan gambar ke buffer.

   Sebelum mengirim event input_image_buffer.append, Anda harus mengirim setidaknya satu event input_audio_buffer.append.

3. Server mendeteksi akhir ucapan dan mengirim event input_audio_buffer.speech_stopped.

4. Server mengirim event input_audio_buffer.committed untuk commit buffer audio.

5. Server mengirim event conversation.item.created yang berisi item pesan pengguna yang dibuat dari buffer.

Mode manual

Atur session.turn_detection ke null dalam event session.update untuk mengaktifkan Mode Manual. Dalam mode ini, client secara eksplisit mengirim event input_audio_buffer.commit dan response.create untuk meminta respons server. Mode ini cocok untuk skenario press-to-talk, seperti mengirim pesan suara di aplikasi chat.

Alur interaksinya sebagai berikut:

1. Client dapat mengirim event input_audio_buffer.append dan input_image_buffer.append kapan saja untuk menambahkan audio dan gambar ke buffer.

   Sebelum mengirim event input_image_buffer.append, Anda harus mengirim setidaknya satu event input_audio_buffer.append.

2. Client mengirim event input_audio_buffer.commit untuk mengirim buffer audio dan gambar, memberi sinyal ke server bahwa semua input pengguna (audio dan gambar) untuk giliran saat ini telah dikirim.

3. Server merespons dengan event input_audio_buffer.committed.

4. Client mengirim event response.create, menunggu server mengembalikan output model.

5. Server merespons dengan event conversation.item.created.

qwen3.5-omni-realtime

qwen3-omni-flash-realtime-2025-12-01

qwen3-omni-flash-realtime, qwen3-omni-flash-realtime-2025-09-15

Qwen-Omni-Turbo-Realtime