Add L16 linear PCM codec for AudioBridge media - #2
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Support l16 and l16-48 on join, decode/encode, plain RTP and rtpws call_info with room sampling_rate validation; extend ws-audio-test. Co-authored-by: Cursor <cursoragent@cursor.com>
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Superseded by #7, which carries this same L16 commit plus the payload-only framing feature (stacked on feature/audiobridge-rtpws). Closing to avoid duplicate L16 commits in the stack. |
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Summary
l16@ 16 kHz, PT 106;l16-48@ 48 kHz, PT 105) on plain RTP andmedia=websocketjoins.call_infoto the participant join codec (not hardcoded Opus).sampling_rate.ws-audio-testwith--codec l16/l16-48and BE PCM tone generation.Base:
master(fork-only draft, not targeting meetecho/janus-gateway).Usage
Expected
call_info:{"type":"call_info","codec":"l16","sample_rate":16000,"channels":1,"ptime_ms":20,"payload_type":106}Test plan
codec=l16in a 16 kHz room →call_infoPT 106, 640-byte payloadscodec=l16-48in a 48 kHz room →call_infoPT 105, 1920-byte payloadsl16join when roomsampling_rateis not 16000ws-stream --codec l16 --tone --expect-rx 50bidirectional RTP