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Audio Capture for .NET Developers
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Introduction to Audio Capture
Our SDK provides robust audio capture capabilities designed specifically for .NET developers. Whether you're building a professional recording application, adding voice chat to your software, or creating a podcasting tool, our audio capture components deliver exceptional performance and flexibility.
The audio capture functionality lets you record from any audio input device on the system, including microphones, line-in ports, and virtual audio devices. All processing is optimized for minimal CPU usage while maintaining pristine audio quality.
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Supported Audio Sources
The SDK supports capturing from multiple audio sources:
- Physical microphones - Desktop, USB, and Bluetooth microphones
- Line-in ports - For capturing from external mixers or instruments
- Virtual audio devices - Capture audio from other applications
- System audio - Record what's playing through your speakers
- Network streams - Capture audio from RTSP, HTTP, and other streaming sources
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Audio Format Support
Our SDK allows you to capture and encode audio in various industry-standard formats to meet any requirement:
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Lossy Formats
- MP3 - Industry standard compressed audio with adjustable bitrates from 8kbps to 320kbps
- M4A (AAC) - Advanced Audio Coding with excellent quality-to-size ratio
- Windows Media Audio - Microsoft's audio format with good compression and Windows integration
- Ogg Vorbis - Free and open-source format with excellent quality at lower bitrates
- Speex - Optimized for speech with good quality at very low bitrates
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Lossless Formats
- WAV - Uncompressed audio with perfect quality and wide compatibility
- FLAC - Free Lossless Audio Codec providing compression without quality loss
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Key Features
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Device Control
- Enumerate all available audio input devices
- Select specific input devices programmatically
- Set input volume levels and mute status
- Monitor audio levels in real-time
- Auto-select default system devices
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Advanced Processing
- Real-time audio visualization with spectrum and waveform analysis
- Noise reduction and echo cancellation
- Gain control and normalization
- Voice activity detection (VAD)
- Stereo/mono channel management
- Sample rate conversion
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Recording Controls
- Start, pause, resume, and stop recording
- Buffer management for low-latency operation
- Timed recordings with automatic stop
- File splitting for large recordings
- Auto file naming with timestamps
- Recording profiles for quick setup
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Best Practices
For optimal audio capture in your applications:
- Always check device availability before starting capture
- Monitor audio levels during recording to detect silence or clipping
- Use appropriate formats based on your quality and file size requirements
- Implement error handling for device disconnection events
- Provide visual feedback to users during recording
- Test on various hardware to ensure compatibility
- Apply noise reduction only when needed as it can affect audio quality
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Audio Capture Integration
The audio capture component integrates seamlessly with other SDK elements:
- Combine with video capture for complete AV recording
- Mix with audio playback for call recording applications
- Use with streaming components for live broadcasting
- Integrate with timeline editor for basic audio editing
- Pair with file conversion for post-processing workflows
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Performance Considerations
The SDK is optimized for efficiency, but here are some tips for best performance:
- Lower sample rates (44.1kHz vs 48kHz) reduce CPU usage
- Mono recording uses less processing power than stereo
- MP3 encoding is more CPU-intensive than WAV recording
- Higher bitrates require more processing power
- Buffer sizes affect latency and stability
Visit our GitHub page to get more code samples and implementation examples.