[BUG] SDK audio defaults differ from WebSocket protocol defaults

[MaxMansfield]

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.

Parameter	SDK Default	WebSocket Default
Codec	OPUS (4)	L16 (1)
Sample Rate	48kHz (3)	16kHz (1)
Channel	Stereo (2)	Mono (1)
Data Option	AUDIO_MULTI_STREAMS (2)	AUDIO_MIXED_STREAM (1)
Duration	20ms	20ms
Frame Size	960	320

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