The Tool Desk
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Start with the streaming path
“Live streaming” can mean interactive audio between browsers, sending a contribution stream to a platform, or packaging a stream for delivery to viewers. Those workflows have different codec and packaging requirements. Before changing an encoder, identify the protocol, destination, channel layout, and playback endpoints.
| Workflow | Practical starting point | What to verify |
|---|---|---|
| Browser-to-browser real-time audio over WebRTC | Opus; retain G.711 support where fallback interoperability is needed. | Endpoint support, speech versus music, channel count, bandwidth, and latency. MDN’s WebRTC codec guide says WebRTC-compatible browsers must support Opus and G.711 PCMA/PCMU. |
| YouTube Live ingest over RTMP/RTMPS | AAC is a safe choice from YouTube’s listed formats; MP3 is also listed for general audio. | For 5.1 surround, YouTube’s general encoder guidance specifies AAC. Check settings for the selected protocol. YouTube’s live encoder settings list accepted formats. |
| YouTube Live ingest over HLS | AAC, following the HLS ingest requirements. | HLS settings are not interchangeable with RTMP settings. YouTube’s HLS setup guidance recommends 128 kbps stereo or 384 kbps 5.1. See YouTube’s HLS setup guidance and Google’s HLS delivery documentation. |
| HLS delivery targeting Apple devices | AAC-LC or another explicitly supported option appropriate to the target and channel layout. | Check Apple’s current codec, container, channel, and bitrate requirements in its HLS authoring specification. |
When to choose Opus
Opus is a strong option for interactive, low-latency WebRTC audio. MDN identifies it as the primary WebRTC audio format and recommends considering it for audio above 8 kHz. Its bitrate recommendations vary with content and channel count; they are not universal targets for platform ingest.
At a 20 ms frame size, MDN gives these recommended Opus/WebRTC ranges:
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| Content | Recommended bitrate |
|---|---|
| Narrow-band speech | 8–12 kbps |
| Wide-band speech | 16–20 kbps |
| Full-band speech | 28–40 kbps |
| Full-band monaural music | 48–64 kbps |
| Full-band stereo music | 64–128 kbps |
These figures are for Opus in the WebRTC context. They should not be copied directly into a YouTube encoder preset without checking that workflow’s requirements. WebRTC codec requirements also do not establish identical support in HTML media elements; browser playback support depends on the media context. MDN’s separate web audio codec guide covers codec and container combinations.
When to choose AAC
AAC is a practical choice when the destination specifies it, including YouTube HLS ingest and YouTube’s documented 5.1 audio support for general RTMP/RTMPS live encoder guidance. AAC-family formats are also documented for Apple HLS authoring. Confirm the exact profile, packaging, channel layout, and bitrate for the target rather than assuming that all AAC configurations are equivalent.
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Apple’s authoring guidance recommends 32–160 kbit/s total for stereo AAC and gives different ranges for other formats and channel layouts. Treat that as Apple HLS authoring guidance, not a universal preset for every live platform.
What bitrate should you use?
Use the number published for your actual workflow, not the largest or most familiar figure. The WebRTC Opus ranges above depend on speech or music and mono or stereo. YouTube’s HLS setup figures—128 kbps stereo or 384 kbps 5.1—are recommendations for that ingest workflow. Apple’s stereo AAC range applies to Apple HLS authoring. These answer different configuration questions.
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- Match the bitrate recommendation to the protocol and destination.
- Choose mono, stereo, or surround deliberately; channel layout changes bandwidth needs and compatibility.
- Leave network headroom for the full stream, including video and protocol overhead.
- Check that the encoder and playback endpoints support the same codec and configuration.
AAC or Opus: which should you use?
For interactive browser audio over WebRTC, start with Opus. For YouTube platform ingest, follow YouTube’s format guidance: AAC is suitable from the listed RTMP/RTMPS choices, while the HLS ingest guidance specifies AAC. For HLS delivery aimed at Apple devices, use an option that meets Apple’s authoring specification. The destination and protocol—not a general codec ranking—decide.
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If your goal is a continuous YouTube stream built from uploaded recordings rather than an interactive WebRTC session, codec choice is only one part of the setup. YouTube must accept the ingest format, and you must have the rights to stream the recording. A codec setting does not grant permission to use music or footage, and platform eligibility or monetization decisions are separate from technical compatibility.
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