[BUG] SDK audio defaults differ from WebSocket protocol defaults #92

Bug Description

The SDK uses different default audio parameters than the raw RTMS WebSocket protocol. While the SDK defaults provide better quality (OPUS, 48kHz, Stereo, AUDIO_MULTI_STREAMS), this can cause confusion when users expect the WebSocket defaults.

Proposed Solution for v2.0

Align SDK defaults with WebSocket protocol defaults for consistency:

• Codec: L16 (1)
• Sample Rate: 16kHz (1)
• Channel: Mono (1)
• Data Option: AUDIO_MIXED_STREAM (1)
• Frame Size: 320

Users who want the current SDK defaults (higher quality, per-speaker attribution) would need to explicitly configure them.

Workaround (v1.x)

Users can configure audio params to match WebSocket defaults:

client.setAudioParams({
    contentType: rtms.AudioContentType.RAW_AUDIO,
    codec: rtms.AudioCodec.L16,
    sampleRate: rtms.AudioSampleRate.SR_16K,
    channel: rtms.AudioChannel.MONO,
    dataOpt: rtms.AudioDataOption.AUDIO_MIXED_STREAM,
    duration: 20,
    frameSize: 320
});

Language/Runtime

Node.js and Python

SDK Version

1.0.x

Additional Context

This is documented in the SDK documentation with examples for both configurations.

Verification

• I've searched existing issues to ensure this bug hasn't already been reported
• I've verified this bug still exists in the latest version of the SDK
• I've included all necessary information to reproduce this issue