Audio Streaming
Learn how to stream and buffer audio data
Overview
flutter_soloud supports streaming audio data while receiving it in real-time. The supported audio data formats are raw PCM, MP3, or compressed through the Opus and Vorbis codecs with Ogg container libraries from Xiph.org. This is particularly useful when:
- Streaming audio from network sources like icecast radio
- Working with OpenAI or other streaming APIs
- Generating audio data on-the-fly
- Processing audio in chunks
- Automatically pause when buffering is needed and resume playback when enough data is available
The Opus and Ogg libraries are embedded by default in flutter_soloud. However, if you don't need streaming capabilities, please read the Without Opus/Ogg section for how to exclude these libraries from your app.
Buffer Stream Setup
Initialize an audio stream:
final stream = SoLoud.instance.setBufferStream(
maxBufferSizeBytes: 1024 * 1024 * 10, // 10MB of max buffer (not allocated)
bufferingType: BufferingType.preserved, // Keep all data in memory
bufferingTimeNeeds: 2.0, // 2 seconds to buffer before unpausing
sampleRate: 44100, // 44100 Hz
channels: Channels.stereo, // 2 channels
format: BufferType.auto, // Autodetect format (not valid for raw PCM)
onBuffering: (isBuffering, handle, time) {
// Handle buffering
},
onMetadata: (metadata) {
// Handle metadata
},
);
Parameters:
| Parameter | Description |
|---|---|
maxBufferSizeBytes | Maximum buffer size in bytes. When this limit is reached while adding audio data, the stream is considered ended (as if setDataIsEnded was called). Playback will stop at this point unless looping is enabled. Internally, all data is stored as floats, regardless of input format. This does not allocate memory upfront; it only limits the total data that can be added. |
maxBufferSizeDuration | Alternative to maxBufferSizeBytes, specifies the maximum buffer size as a duration (in seconds), calculated using sampleRate and channels. No memory is allocated upfront. |
bufferingType | Determines how buffering works during playback. See below.BufferingType.preserved: Keeps all audio data in memory, allows multiple playback instances, supports seeking and looping. BufferingType.released: Frees memory of already played data, allows only a single playback instance, and must be manually disposed. |
bufferingTimeNeeds | Buffering time required (in seconds). If playback reaches the end of the current buffer, it will pause and wait until enough data is buffered to cover this time. Note: With BufferingType.released, the stream position is always 0; use getStreamTimeConsumed to get elapsed time. |
sampleRate | Sample rate for playback (e.g., 22050 or 44100 Hz). For opus format, valid values are 48000, 24000, 16000, 12000, or 8000 Hz. Incoming data is resampled to this rate. |
channels | Number of audio channels. The opus format supports only mono and stereo. |
format | Audio data format. Options: f32le, s8, s16le, s32le, orauto. Note: the auto autodetect MP3 and Ogg container with Opus or Vorbis. With this format, the samplerate and channels parameters are ignored. |
onBuffering | Callback triggered when buffering starts (isBuffering = true) and ends (isBuffering = false). Receives the playback handle and the current buffered time (in seconds). |
onMetadata | Callback triggered when starting to add audio data or when metadata changes while streaming. It returns a AudioMetadata object. |
Preserved Mode
final stream = await SoLoud.instance.setBufferStream(
bufferingType: BufferingType.preserved,
// ...other parameters
);
- Keeps all audio data in memory
- Allows multiple playback instances
- Supports seeking and looping
- Higher memory usage
Released Mode
final stream = await SoLoud.instance.setBufferStream(
bufferingType: BufferingType.released,
// ...other parameters
);
- Frees played audio data
- The position of the stream is always 0
- The seek method is not supported
- Single playback instance only
- Lower memory usage
- Must be manually disposed
WARNING: as you can see, the position of the stream in released mode is always at start. This means that getPosition always returns 0. To get the already played time, you should use the getStreamTimeConsumed method instead (basically it is the current position). Also, seek is not supported in released mode.
Please, look at the example/lib/buffer_stream/simple_noise_stream.dart example for a simple implementation to understand how audio stream works.
Supported Formats
For raw PCM data, the following formats are supported:
s8- Signed 8-bit PCMs16le- Signed 16-bit PCM (little endian)s32le- Signed 32-bit PCM (little endian)f32le- 32-bit float PCM (little endian)
You should also need to specify the sample rate and the number of channels.
auto- Automatically detect the format
Autodetected formats are:
mp3- Mp3 formatopus- Ogg container with Opus codecogg- Ogg container with Vorbis codec These compressed formats are supporting metadata (using theonMetadatacallback).
Adding Audio Data
// Add audio data to the stream
SoLoud.instance.addAudioDataStream(
stream,
audioChunk, // Uint8List of audio data
);
// Mark the stream as complete
SoLoud.instance.setDataIsEnded(stream);
Buffer Management
// Get current buffer size in bytes
final size = SoLoud.instance.getBufferSize(stream);
// Reset the buffer
SoLoud.instance.resetBufferStream(stream);
Example: Network Streaming
// Create a WebSocket connection
final socket =
await WebSocket.connect('wss://audio-stream.example.com');
// Set up the audio stream
final stream = SoLoud.instance.setBufferStream(
bufferingType: BufferingType.released,
format: BufferType.opus,
onBuffering: (isBuffering, handle, time) {
// When isBuffering==true, the stream is set to paused automatically till
// it reaches bufferingTimeNeeds of audio data or until setDataIsEnded is called
// or maxBufferSizeBytes is reached. When isBuffering==false, the playback stream
// is resumed.
print('Buffering: $isBuffering, Time: $time');
},
onMetadata: (metadata) {
debugPrint(metadata.toString());
},
}
);
// Start the the playback whenever enough data is buffered (defined by `bufferingTimeNeeds` seconds) whenever enough data is buffered (defined by `bufferingTimeNeeds` seconds)
final handle = await SoLoud.instance.play(stream);
// Eventually listen for the handle to finish playing.
// `SoLoud.setDataIsEnded` must be called before this listener
// could be emitted.
stream.soundEvents.listen((event) {
if (event.event == SoundEventType.handleIsNoMoreValid &&
event.handle == handle) {
print('sound has finished playing');
}
});
// Listen for audio data
socket.listen(
(data) {
if (data is List<int>) {
SoLoud.instance.addAudioDataStream(
stream,
Uint8List.fromList(data),
);
}
},
onDone: () {
// Mark the stream as completed.
// You can then listen to the `stream.soundEvents` events
SoLoud.instance.setDataIsEnded(stream);
},
);
Please, look at the example/lib/buffer_stream/web_radio.dart example for a simple implementation to connect to an online icecast web radio or at example/lib/buffer_stream/websocket.dart example to connect to a WebSocket server (please read the note at the beginning).
Example: PCM Generation
@pragma('vm:entry-point')
Future<AudioSource> generatePCM() async {
final pcmStream = SoLoud.instance.setBufferStream(
maxBufferSizeBytes: 1024 * 1024,
format: BufferType.s16le,
channels: Channels.mono,
sampleRate: 44100,
);
// Generate some PCM data
final buffer = Int16List(44100);
for (var i = 0; i < buffer.length; i++) {
buffer[i] = (sin(2 * pi * 440 * i / 44100) * 32767).toInt();
}
// Add to stream
SoLoud.instance.addAudioDataStream(
pcmStream,
buffer.buffer.asUint8List(),
);
SoLoud.instance.setDataIsEnded(pcmStream);
return pcmStream;
}