1.Offline speech to text (ASR)

1. Introduction

1.1 What is "ASR"?

ASR (Automatic Speech Recognition) is a technology that converts human speech signals into text. It is widely used in intelligent assistants, voice command control, telephone customer service automation, and real-time subtitle generation. The goal of ASR is to enable machines to "understand" human speech and convert it into a form that computers can process and understand.

1.2 Implementation Principles

The implementation of an ASR system relies primarily on the following key technical components:

1. Acoustic Model

• The acoustic model is responsible for converting the input audio signal into phonemes or subword units. This typically involves a feature extraction step, such as using Mel-Frequency Cepstral Coefficients (MFCCs) or filter banks to represent the audio signal.
• These features are then fed into a deep neural network (DNN), convolutional neural network (CNN), recurrent neural network (RNN), or the more advanced Transformer architecture for training to learn how to map audio features to corresponding phonemes or text.

2. Language Model

• A language model predicts the most likely next word given the previous context, thereby improving recognition accuracy. It is trained on large amounts of text data to understand which word sequences are most likely to occur.
• Common language models include n-gram models, RNN-based language models (LMs), and the more recently popular Transformer-based language models.

3. Pronunciation Dictionary

• A pronunciation dictionary provides a mapping between words and their corresponding pronunciations. This is crucial for connecting the acoustic model and language model, as it allows the system to understand and match the sounds it hears based on known pronunciation rules.

4. Decoder

• The decoder's task is to find the most likely word sequence as output given the acoustic model, language model, and pronunciation dictionary. This process typically involves complex search algorithms, such as the Viterbi algorithm or graph-based search methods, to find the optimal path.

5. End-to-End ASR

• With the advancement of deep learning, end-to-end ASR systems have emerged. These systems attempt to learn text output directly from raw audio signals, without requiring explicit acoustic models, separate pronunciation lexicons, and language models. These systems are often based on sequence-to-sequence (Seq2Seq) frameworks, using, for example, attention mechanisms or the Transformer architecture, significantly simplifying the complexity of traditional ASR systems.

In general, modern ASR systems achieve efficient and accurate speech-to-text conversion by combining the aforementioned components and leveraging large datasets and powerful computing resources for training. With technological advances, the performance of ASR systems continues to improve, and their application scenarios are becoming increasingly broad.

2. Code Analysis

Key Code

1. Speech Processing and Recognition Core (largemodel/largemodel/asr.py)

xxxxxxxxxx
# From largemodel/largemodel/asr.py
def kws_handler(self)->None:
    if self.stop_event.is_set():
        return      
​
    if self.listen_for_speech(self.mic_index):
        asr_text = self.ASR_conversion(self.user_speechdir)  # 进行 ASR 转换 / Perform ASR conversion
        if asr_text =='error':  # 检查 ASR 结果长度是否小于4个字符 / Check if ASR result length is less than 4 characters
            self.get_logger().warn("I still don't understand what you mean. Please try again")
            playsound(self.audio_dict[self.error_response])  # 错误响应 / Error response
        else:
            self.get_logger().info(asr_text)
            self.get_logger().info("okay😀, let me think for a moment...")
            self.asr_pub_result(asr_text)  # 发布 ASR结果 / Publish ASR result            
    else:
        return
​
def ASR_conversion(self, input_file:str)->str:
    if self.use_oline_asr:
        result=self.modelinterface.oline_asr(input_file)
        if result[0] == 'ok' and len(result[1]) > 4:
            return result[1]
        else:
            self.get_logger().error(f'ASR Error:{result[1]}')  # ASR error.
            return 'error'
    else:
        result=self.modelinterface.SenseVoiceSmall_ASR(input_file)
        if result[0] == 'ok' and len(result[1]) > 4:
            return result[1]
        else:
            self.get_logger().error(f'ASR Error:{result[1]}')  # ASR error.
            return 'error'

2. VAD smart recording (largemodel/largemodel/asr.py)

xxxxxxxxxx
# From largemodel/largemodel/asr.py
def listen_for_speech(self,mic_index=0):
    p = pyaudio.PyAudio()   # Create PyAudio instance. / 创建PyAudio实例。
    audio_buffer = []       # Store audio data. / 存储音频数据。
    silence_counter = 0     # Silence counter. / 静音计数器。
    MAX_SILENCE_FRAMES = 90  # 30帧*30ms=900ms静音后停止 / Stop after 900ms of silence (30 frames * 30ms)
    speaking = False  # Flag indicating speech activity. / 语音活动标志。
    frame_counter = 0  # Frame counter. / 计数器。
    stream_kwargs = {
        'format': pyaudio.paInt16,
        'channels': 1,
        'rate': self.sample_rate,
        'input': True,
        'frames_per_buffer': self.frame_bytes,
    }
    if mic_index != 0:
        stream_kwargs['input_device_index'] = mic_index
​
    # Prompt the user to speak via the buzzer. / 通过蜂鸣器提示用户讲话。
    self.pub_beep.publish(UInt16(data = 1))
    time.sleep(0.5)
    self.pub_beep.publish(UInt16(data = 0))
​
    try:
        # Open audio stream. / 打开音频流。
        stream = p.open(**stream_kwargs)
        while True:
            if self.stop_event.is_set():
                return False 
    
            frame = stream.read(self.frame_bytes, exception_on_overflow=False)  # Read audio data. / 读取音频数据。
            is_speech = self.vad.is_speech(frame, self.sample_rate)  # VAD detection. / VAD检测。
​
            if is_speech:
                # Detected speech activity. / 检测到语音活动。
                speaking = True
                audio_buffer.append(frame)
                silence_counter = 0
            else:
                if speaking:
                    # Detect silence after speech activity. / 在语音活动后检测静音。
                    silence_counter += 1
                    audio_buffer.append(frame)  # Continue recording buffer. / 持续记录缓冲。
                    
                    # End recording when silence duration meets the threshold. / 静音持续时间达标时结束录音。
                    if silence_counter >= MAX_SILENCE_FRAMES:
                        break
            frame_counter += 1
            if frame_counter % 2 == 0:
                self.get_logger().info('1' if is_speech else '-')
                # Real-time status display.
    finally:
        stream.stop_stream()
        stream.close()
        p.terminate()
​
    # Save valid recording (remove trailing silence). / 保存有效录音（去除尾部静音）。
    if speaking and len(audio_buffer) > 0:
        # Trim the last silent part. / 裁剪最后静音部分。
        clean_buffer = audio_buffer[:-MAX_SILENCE_FRAMES] if len(audio_buffer) > MAX_SILENCE_FRAMES else audio_buffer
        
        with wave.open(self.user_speechdir, 'wb') as wf:
            wf.setnchannels(1)
            wf.setsampwidth(p.get_sample_size(pyaudio.paInt16))
            wf.setframerate(self.sample_rate)
            wf.writeframes(b''.join(clean_buffer))
            return True

Code Analysis

ASR (speech-to-text) functionality is provided by the ASRNode node (asr.py). This node is responsible for recording, converting, and publishing audio.

1. Audio Recording (listen_for_speech):

• This function uses the pyaudio library to capture the audio stream from the microphone.
• It integrates the webrtcvad library for voice activity detection (VAD). The function loops through audio frames and uses vad.is_speech() to determine whether each frame contains human voice.
• When speech is detected, the data is written to a buffer. Recording stops when sustained silence (defined by MAX_SILENCE_FRAMES) is detected.
• Finally, the audio data in the buffer is written to a .wav file in the self.user_speechdir directory.

2. Backend Selection and Execution (ASR_conversion):

• The kws_handler function calls the ASR_conversion function after successful recording.
• This function determines which backend implementation to call by reading the ROS parameter use_oline_asr (a Boolean value).
• If false, the self.modelinterface.SenseVoiceSmall_ASR method is called for local recognition.
• If true, the self.modelinterface.oline_asr method is called for online recognition.
• This function passes the audio file path as a parameter to the selected method and processes the returned result.

3. Result Publishing (asr_pub_result):

• After ASR_conversion
  Voice interaction hardware connection (ReadMe)

  Offline text to speech (TTS)