Audio to Stream

A Windows application that captures audio from any input device and streams it in real-time using the SRT (Secure Reliable Transport) protocol. Perfect for streaming audio to OBS Studio or other media applications with low latency.

Overview

Audio to Stream provides a simple GUI to:

• Select any audio input device (microphone, line-in, WASAPI loopback, etc.)
• Configure streaming bitrate (64k - 320k)
• Stream audio via SRT protocol with customizable URLs
• Monitor audio levels with real-time VU meters
• Track streaming statistics (bitrate, connection status, duration)

The application uses FFmpeg for encoding and streaming, with SRT protocol ensuring reliable transmission even over unreliable networks.

Features

• 🎵 Multiple Audio Sources: Support for all Windows audio input devices
• 🎚️ Adjustable Bitrate: Choose from 64k to 320k for quality vs. bandwidth tradeoffs
• 📊 Real-time VU Meters: Monitor left and right channel audio levels
• 📡 SRT Streaming: Low-latency, reliable streaming protocol
• 💾 Persistent Settings: Automatically saves your configuration
• 🎨 Dark Theme UI: Easy on the eyes during long streaming sessions
• 📈 Live Statistics: View connection status, bitrate, and streaming duration

Requirements

• Windows OS
• FFmpeg (bundled with the executable or available in PATH)
• Audio input device

Build in Python v3.10.10

Usage

Basic Setup

1. Launch Audio to Stream
2. Select your audio input device from the dropdown
3. Choose your desired bitrate (192k recommended for good quality)
4. Configure the stream URL (default: srt://localhost:9000)
5. Click Start Streaming

Stream URL Format

The default SRT URL format is:

srt://<ip_address>:<port>

For example:

• srt://localhost:9000 - Stream locally
• srt://192.168.1.100:9000 - Stream to another device on your network
• srt://0.0.0.0:9000?mode=listener - Listen mode (wait for connection)

OBS Studio Integration

To receive the audio stream in OBS Studio, follow these steps:

Adding an SRT Media Source

1. Add a Media Source

   • In OBS Studio, click the + button in the Sources panel
   • Select Media Source
   • Give it a name (e.g., "Audio Stream")
   • Click OK

2. Configure the Media Source

   • Uncheck "Local File"

   • In the Input field, enter your SRT URL with parameters:

     srt://<local_ip>:<port>?mode=listener&latency=2000000
     

     Example:

     srt://192.168.1.100:9000?mode=listener&latency=2000000
     

     Or for localhost:

     srt://127.0.0.1:9000?mode=listener&latency=2000000
     

3. Set Input Format

   • In the Input Format field, enter:

     mpegts
     

4. Configure Network Options

   • Uncheck "Buffering"
   • Set Network Buffering to 0 MB or uncheck the buffering option entirely
   • Set Reconnect Delay to 2 seconds (2000 ms)

5. Apply Settings

   • Click OK to save the media source
   • The audio should now appear in OBS

Connection Flow

The typical setup uses:

• Audio to Stream: Runs in caller mode (connects to OBS)
• OBS Studio: Runs in listener mode (waits for the connection)

Make sure to:

1. Start OBS Studio first and add the media source
2. Then start streaming from Audio to Stream

SRT URL Parameters Explained

• mode=listener - OBS waits for incoming connections (recommended for OBS side)
• latency=2000000 - Sets latency to 2000ms (2 seconds) in microseconds. Adjust based on network conditions:
  • LAN/Local: 1000000 (1 second)
  • Stable network: 2000000 (2 seconds)
  • Unreliable network: 3000000-5000000 (3-5 seconds)

Troubleshooting OBS Connection

If you don't see/hear audio in OBS:

1. Check IP and Port: Ensure the IP address and port match between Audio to Stream and OBS
2. Firewall: Windows Firewall may block SRT connections. Allow OBS through the firewall
3. Start Order: Start OBS's media source (listener) before starting Audio to Stream (caller)
4. Test Locally First: Use 127.0.0.1 or localhost to test on the same machine
5. Check Audio Device: Verify the correct audio device is selected in Audio to Stream
6. Monitor VU Meters: Ensure audio levels are showing in Audio to Stream
7. Increase Latency: If stream is choppy, try increasing the latency parameter

Configuration File

Settings are automatically saved to settings.ini in the application directory:

[Settings]
audio_device = 0
bitrate = 192k
stream_url = srt://localhost:9000

Technical Details

• Encoding: AAC audio codec
• Container: MPEGTS (MPEG Transport Stream)
• Protocol: SRT (Secure Reliable Transport)
• Sample Rate: 44100 Hz (configurable via FFmpeg)
• Channels: Stereo (2 channels)

Building from Source

See BUILD_INSTRUCTIONS.md for details on building the executable with PyInstaller.

Dependencies

Install required Python packages:

pip install -r requirements.txt

Required packages:

• sounddevice - Audio device interface
• numpy - Audio data processing
• PyInstaller - Executable building
• pillow - Image processing (for icon)

License

This project is provided as-is for personal and commercial use.

Credits

• Uses FFmpeg for audio encoding and streaming
• Built with Python and tkinter
• SRT protocol implementation via FFmpeg

Support

For issues, questions, or contributions, please visit the project repository on GitHub.

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Note: Ensure FFmpeg is available either bundled with the application or installed in your system PATH.