Audio Requirements
Required Format Raw PCM, 16,000 Hz, 16-bit Signed Little-Endian, Mono
Technical Specifications Parameter Value Description Format PCM (Pulse Code Modulation) Uncompressed linear audio Sample Rate 16,000 Hz 16,000 samples per second Bit Depth 16-bit Signed 16-bit integers Channels 1 (Mono) Single audio channel Endianness Little-Endian Least significant byte first Encoding Signed Integer Two’s complement representation Byte Order Low byte, High byte For each 16-bit sample
16-bit Signed Integer Range Value Hex Binary Description 327670x7FFF0111111111111111Maximum positive 10x00010000000000000001Smallest positive 00x00000000000000000000Digital silence -10xFFFF1111111111111111Smallest negative -327680x80001000000000000000Maximum negative
Little-Endian Byte Order For a 16-bit sample value of 0x1234:
Low byte (LSB) : 0x34 (stored first)High byte (MSB) : 0x12 (stored second)
Memory layout: Address: 0x00 0x01 0x02 0x03 Value: 0x34 0x12 0x78 0x56 Sample: ←─0x1234──┘ ←─0x5678──┘ Data Rate Calculations Bandwidth Requirements Sample Rate: 16,000 Hz Bit Depth: 16 bits = 2 bytes Channels: 1 (Mono) Bytes per second = 16,000 × 2 × 1 = 32,000 bytes/sec Bits per second = 32,000 × 8 = 256,000 bps = 256 kbps Buffer Size Recommendations Duration Samples Bytes Use Case 10ms 160 320 Minimal latency 20ms 320 640 Recommended 50ms 800 1,600 Standard buffer 100ms 1,600 3,200 Large buffer
Recommended : 20ms chunks (640 bytes) provide the best balance of latency and reliability.
Implementation Examples // Configure FlutterSound for recording await _recorder. startRecorder ( codec : Codec .pcm16, // 16-bit PCM sampleRate : 16000 , // 16kHz numChannels : 1 , // Mono bitRate : 256000 , // 256 kbps ); // Configure FlutterSound for playback await _player. startPlayerFromStream ( codec : Codec .pcm16, // 16-bit PCM numChannels : 1 , // Mono sampleRate : 16000 , // 16kHz ); Audio Stream Processing: _recorderSubscription = stream ? . listen (( Uint8List buffer) { // Verify buffer is correctly formatted assert (buffer.length % 2 == 0 , 'Buffer must be even length' ); // Send to WebSocket _channel ? .sink. add (buffer); }); Audio Quality Guidelines Recording Best Practices
Microphone Setup Use close-talking microphone 6-12 inches from speaker
Environment Record in quiet environment with minimal background noise
Gain Control Avoid automatic gain control (AGC) for consistent levels
Sample Quality Maintain signal levels between -12dB to -6dB for optimal quality
Audio Processing Chain Validation and Testing function validateAudioFormat ( audioBuffer ) { // Check if buffer length is even (16-bit samples) if ( audioBuffer . byteLength % 2 !== 0 ) { throw new Error ( 'Invalid audio buffer: length must be even' ); } // Check for reasonable buffer size const minSize = 160 ; // 10ms at 16kHz const maxSize = 8000 ; // ~250ms at 16kHz if ( audioBuffer . byteLength < minSize ) { throw new Error ( 'Audio buffer too small' ); } if ( audioBuffer . byteLength > maxSize ) { throw new Error ( 'Audio buffer too large' ); } return true ; } Audio Level Monitoring class AudioLevelMonitor { static double calculateRMS ( Uint8List pcmData) { double sum = 0.0 ; final samples = pcmData.length ~/ 2 ; for ( int i = 0 ; i < pcmData.length; i += 2 ) { // Convert little-endian bytes to 16-bit signed int final sample = (pcmData[i + 1 ] << 8 ) | pcmData[i]; final signedSample = sample > 32767 ? sample - 65536 : sample; sum += signedSample * signedSample; } return sqrt (sum / samples); } static double rmsToDecibels ( double rms) { return 20 * log (rms / 32767 ) / ln10; } } Common Issues and Solutions
Symptoms:
Crackling or popping sounds
Metallic audio quality
Clipped audio
Causes:
Incorrect sample rate conversion
Audio clipping (levels too high)
Wrong endianness
Incorrect bit depth
Solutions: // Ensure proper audio configuration await recorder. startRecorder ( codec : Codec .pcm16, // Correct: 16-bit PCM sampleRate : 16000 , // Correct: 16kHz numChannels : 1 , // Correct: Mono );
Symptoms:
No sound transmitted or received
Zero bytes in audio buffers
WebSocket receives empty data
Debug Steps: // Check audio permissions final status = await Permission .microphone.status; print ( 'Microphone permission: $ status ' ); // Monitor audio data stream ? . listen ((buffer) { print ( 'Audio buffer size: ${ buffer ?. length } bytes' ); if (buffer ? .isNotEmpty == true ) { final level = AudioLevelMonitor . calculateRMS (buffer ! ); print ( 'Audio level: ${ AudioLevelMonitor . rmsToDecibels ( level )} dB' ); } });
Test Audio Generation import numpy as np import wave def generate_test_tone ( frequency = 440 , duration = 1.0 ): """Generate a test tone in correct format""" sample_rate = 16000 samples = int (sample_rate * duration) # Generate sine wave t = np.linspace( 0 , duration, samples, False ) audio = np.sin( 2 * np.pi * frequency * t) # Convert to 16-bit PCM audio_16bit = (audio * 32767 ).astype(np.int16) return audio_16bit.tobytes() # Test with 440Hz tone for 1 second test_audio = generate_test_tone() print ( f "Generated { len (test_audio) } bytes of test audio" ) function verifyAudioFormat ( audioBuffer ) { const view = new DataView ( audioBuffer ); const samples = audioBuffer . byteLength / 2 ; console . log ( `Audio buffer: ${ audioBuffer . byteLength } bytes` ); console . log ( `Sample count: ${ samples } ` ); console . log ( `Duration: ${ samples / 16000 } seconds` ); // Check first few samples for ( let i = 0 ; i < Math . min ( 10 , samples ); i ++ ) { const sample = view . getInt16 ( i * 2 , true ); // true = little-endian console . log ( `Sample ${ i } : ${ sample } ` ); } } 
