/**
 * Audio codec translation layer for bridging between WebRTC and SIP.
 *
 * All actual codec work (Opus, G.722, PCMU, PCMA) is done in Rust via
 * the smartrust bridge. This module provides the RTP-level transcoding
 * interface used by the webrtcbridge.
 */

import { Buffer } from 'node:buffer';
import { transcode, isCodecReady } from './opusbridge.ts';

/** Opus dynamic payload type (standard WebRTC assignment). */
export const OPUS_PT = 111;

export interface IRtpTranscoder {
  /** Transcode an RTP payload. Always async (Rust IPC). */
  payload(data: Buffer): Promise<Buffer>;
  fromPT: number;
  toPT: number;
}

/**
 * Create a transcoder that converts RTP payloads between two codecs.
 * Returns null if the codecs are the same or the Rust bridge isn't ready.
 *
 * @param sessionId - optional Rust codec session for isolated state per call
 */
export function createTranscoder(fromPT: number, toPT: number, sessionId?: string, direction?: string): IRtpTranscoder | null {
  if (fromPT === toPT) return null;
  if (!isCodecReady()) return null;

  return {
    fromPT,
    toPT,
    async payload(data: Buffer): Promise<Buffer> {
      const result = await transcode(data, fromPT, toPT, sessionId, direction);
      return result || Buffer.alloc(0); // return empty on failure — never pass raw codec bytes
    },
  };
}