node-av 3.1.3 → 5.0.0

package/dist/api/utilities/streaming.js

@@ -1,3 +1,7 @@
+import { AVMEDIA_TYPE_VIDEO } from '../../constants/constants.js';
+import { Codec } from '../../lib/codec.js';
+import { FormatContext } from '../../lib/format-context.js';
+import { Rational } from '../../lib/rational.js';
 import { avSdpCreate } from '../../lib/utilities.js';
 /**
  * Streaming protocol utilities.
@@ -7,11 +11,11 @@ import { avSdpCreate } from '../../lib/utilities.js';
  *
  * @example
  * ```typescript
- * import { StreamingUtils, MediaOutput } from 'node-av/api';
+ * import { StreamingUtils, Muxer } from 'node-av/api';
  *
  * // Create RTP outputs
- * const videoOutput = await MediaOutput.create('rtp://127.0.0.1:5004');
- * const audioOutput = await MediaOutput.create('rtp://127.0.0.1:5006');
+ * const videoOutput = await Muxer.open('rtp://127.0.0.1:5004');
+ * const audioOutput = await Muxer.open('rtp://127.0.0.1:5006');
  *
  * // Generate SDP for streaming
  * const sdp = StreamingUtils.createSdp([videoOutput, audioOutput]);
@@ -23,23 +27,24 @@ import { avSdpCreate } from '../../lib/utilities.js';
  */
 export class StreamingUtils {
     /**
-     * Create an SDP (Session Description Protocol) string from media inputs/outputs
+     * Create an SDP (Session Description Protocol) string from demuxer/muxer
      *
      * Generates an SDP description for RTP/RTSP streaming from one or more
-     * configured media inputs/outputs. The inputs/outputs should be configured with RTP
+     * configured demuxer/muxer. The inputs/outputs should be configured with RTP
      * format and have their streams set up before calling this method.
      *
-     * @param inouts - Array of MediaInput or MediaOutput objects configured for RTP
+     * @param contexts - Alternatively, array of FormatContext objects
+     *
      * @returns SDP string if successful, null if failed
      *
      * @example
      * ```typescript
      * // Set up RTP outputs with streams
-     * const output1 = await MediaOutput.create('rtp://239.0.0.1:5004');
-     * await output1.addVideoStream(encoder1);
+     * const output1 = await Muxer.open('rtp://239.0.0.1:5004');
+     * await output1.addStream(encoder1);
      *
-     * const output2 = await MediaOutput.create('rtp://239.0.0.1:5006');
-     * await output2.addAudioStream(encoder2);
+     * const output2 = await Muxer.open('rtp://239.0.0.1:5006');
+     * await output2.addStream(encoder2);
      *
      * // Generate SDP for multicast streaming
      * const sdp = StreamingUtils.createSdp([output1.getFormatContext(), output2.getFormatContext()]);
@@ -55,15 +60,148 @@ export class StreamingUtils {
         }
         return avSdpCreate(contexts);
     }
+    /**
+     * Generate SDP for RTP/SRTP input using FFmpeg's native SDP generator
+     *
+     * Creates an RFC-compliant SDP description using FFmpeg's internal logic.
+     * This ensures correct codec names, clock rates, and formatting for all codecs.
+     *
+     * @param configs - Array of stream configurations
+     *
+     * @param sessionName - Optional session name for the SDP (default: 'RTP Stream')
+     *
+     * @returns SDP string with proper rtpmap and optional crypto attributes
+     *
+     * @example
+     * ```typescript
+     * import { StreamingUtils } from 'node-av/api';
+     * import { AV_CODEC_ID_OPUS, AV_CODEC_ID_H264 } from 'node-av/constants';
+     *
+     * // Single audio stream with SRTP
+     * const sdp = StreamingUtils.createInputSDP([{
+     *   port: 5004,
+     *   codecId: AV_CODEC_ID_OPUS,
+     *   payloadType: 111,
+     *   clockRate: 48000,
+     *   channels: 2,
+     *   srtp: {
+     *     key: Buffer.alloc(16, 0x12),
+     *     salt: Buffer.alloc(14, 0x34)
+     *   }
+     * }]);
+     *
+     * // Multi-stream (video + audio)
+     * const sdp = StreamingUtils.createInputSDP([
+     *   { port: 5006, codecId: AV_CODEC_ID_H264, payloadType: 96, clockRate: 90000 },
+     *   { port: 5004, codecId: AV_CODEC_ID_OPUS, payloadType: 111, clockRate: 48000, channels: 2 }
+     * ], 'My Stream');
+     * ```
+     */
+    static createInputSDP(configs, sessionName = 'RTP Stream') {
+        const contexts = [];
+        try {
+            // Create one FormatContext per stream with individual URL (port)
+            for (const streamConfig of configs) {
+                const codec = Codec.findDecoder(streamConfig.codecId);
+                if (!codec) {
+                    throw new Error(`Codec not found for codec ID: ${streamConfig.codecId}`);
+                }
+                // Create format context with URL containing the port
+                const ctx = new FormatContext();
+                ctx.allocContext();
+                ctx.url = `rtp://127.0.0.1:${streamConfig.port}`;
+                // Create stream with codec parameters
+                const stream = ctx.newStream(null);
+                stream.codecpar.codecId = streamConfig.codecId;
+                stream.codecpar.codecType = codec.type;
+                // Set audio-specific parameters
+                if (codec.type !== AVMEDIA_TYPE_VIDEO) {
+                    if (streamConfig.clockRate) {
+                        stream.codecpar.sampleRate = streamConfig.clockRate;
+                    }
+                    if (streamConfig.channels) {
+                        stream.codecpar.channels = streamConfig.channels;
+                    }
+                }
+                // Set time base (required for SDP generation)
+                stream.timeBase = new Rational(1, streamConfig.clockRate);
+                contexts.push(ctx);
+            }
+            // Generate SDP using FFmpeg's native generator
+            // FFmpeg will automatically use the port from each context's URL and generate RFC-compliant codec names
+            let sdp = avSdpCreate(contexts);
+            if (!sdp) {
+                throw new Error('Failed to generate SDP');
+            }
+            // Post-process: Replace session name if provided
+            if (sessionName !== 'RTP Stream') {
+                sdp = sdp.replace(/^s=.*$/m, `s=${sessionName}`);
+            }
+            // Post-process: Replace payload types, add SRTP crypto lines, and/or custom fmtp
+            // Note: We always need to process because FFmpeg assigns payload types automatically
+            const sdpLines = sdp.split('\n');
+            const newLines = [];
+            let streamIndex = -1;
+            for (let i = 0; i < sdpLines.length; i++) {
+                const line = sdpLines[i];
+                if (line.startsWith('m=')) {
+                    // New media section
+                    streamIndex++;
+                    const streamCfg = configs[streamIndex];
+                    // Replace FFmpeg's auto-assigned payload type with user-specified one
+                    const replaced = line.replace(/RTP\/AVP\s+(\d+)/, `RTP/AVP ${streamCfg.payloadType}`);
+                    newLines.push(replaced);
+                    // Add SRTP crypto line right after m= line if configured
+                    if (streamCfg?.srtp) {
+                        const suite = streamCfg.srtp.suite ?? 'AES_CM_128_HMAC_SHA1_80';
+                        const keyMaterial = Buffer.concat([streamCfg.srtp.key, streamCfg.srtp.salt]).toString('base64');
+                        newLines.push(`a=crypto:1 ${suite} inline:${keyMaterial}`);
+                    }
+                }
+                else if (line.startsWith('a=rtpmap:')) {
+                    // Replace payload type in rtpmap line
+                    const streamCfg = configs[streamIndex];
+                    const replaced = line.replace(/^a=rtpmap:(\d+)/, `a=rtpmap:${streamCfg.payloadType}`);
+                    newLines.push(replaced);
+                    // If custom fmtp is provided but FFmpeg didn't generate one, add it
+                    if (streamCfg?.fmtp && !sdpLines[i + 1]?.startsWith('a=fmtp:')) {
+                        newLines.push(`a=fmtp:${streamCfg.payloadType} ${streamCfg.fmtp}`);
+                    }
+                }
+                else if (line.startsWith('a=fmtp:')) {
+                    // Replace payload type in fmtp line and optionally replace content
+                    const streamCfg = configs[streamIndex];
+                    if (streamCfg?.fmtp) {
+                        newLines.push(`a=fmtp:${streamCfg.payloadType} ${streamCfg.fmtp}`);
+                    }
+                    else {
+                        const replaced = line.replace(/^a=fmtp:(\d+)/, `a=fmtp:${streamCfg.payloadType}`);
+                        newLines.push(replaced);
+                    }
+                }
+                else {
+                    newLines.push(line);
+                }
+            }
+            return newLines.join('\n');
+        }
+        finally {
+            // Cleanup all contexts
+            for (const ctx of contexts) {
+                ctx.freeContext();
+            }
+        }
+    }
     /**
      * Validate if an output is configured for RTP streaming
      *
-     * @param output - MediaOutput to check
+     * @param output - Muxer to check
+     *
      * @returns true if configured for RTP
      *
      * @example
      * ```typescript
-     * const output = await MediaOutput.create('rtp://127.0.0.1:5004');
+     * const output = await Muxer.open('rtp://127.0.0.1:5004');
      * if (StreamingUtils.isRtpOutput(output)) {
      *   const sdp = StreamingUtils.createSdpForOutput(output);
      * }
@@ -85,8 +223,19 @@ export class StreamingUtils {
      * Helper to construct RTP URLs with proper formatting.
      *
      * @param host - IP address or hostname
+     *
      * @param port - Port number
+     *
      * @param options - Additional options
+     *
+     * @param options.ttl - Time-to-live for multicast
+     *
+     * @param options.localrtpport - Local RTP port
+     *
+     * @param options.localrtcpport - Local RTCP port
+     *
+     * @param options.pkt_size - Packet size
+     *
      * @returns Formatted RTP URL
      *
      * @example
@@ -116,5 +265,45 @@ export class StreamingUtils {
         }
         return url;
     }
+    /**
+     * Extract all UDP ports from SDP content
+     *
+     * @param sdp - SDP content string
+     *
+     * @returns Array of port numbers (one per stream)
+     *
+     * @example
+     * ```typescript
+     * import { StreamingUtils } from 'node-av/api';
+     *
+     * const sdp = `v=0
+     * o=- 0 0 IN IP4 127.0.0.1
+     * s=Test Stream
+     * c=IN IP4 127.0.0.1
+     * t=0 0
+     * m=audio 5004 RTP/AVP 111
+     * a=rtpmap:111 OPUS/48000/2
+     * m=video 5006 RTP/AVP 96
+     * a=rtpmap:96 H264/90000`;
+     *
+     * const ports = StreamingUtils.extractPortsFromSDP(sdp);
+     * console.log(ports); // [5004, 5006]
+     * ```
+     */
+    static extractPortsFromSDP(sdp) {
+        const ports = [];
+        const lines = sdp.split('\n');
+        for (const line of lines) {
+            if (line.startsWith('m=')) {
+                // m=audio 5004 RTP/AVP 111
+                // m=video 5006 RTP/AVP 96
+                const match = /m=\w+\s+(\d+)/.exec(line);
+                if (match) {
+                    ports.push(parseInt(match[1], 10));
+                }
+            }
+        }
+        return ports;
+    }
 }
 //# sourceMappingURL=streaming.js.map

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package/dist/api/utilities/timestamp.d.ts

@@ -1,4 +1,4 @@
-import type { AVRounding } from '../../constants/constants.js';
+import type { AVRounding } from '../../constants/index.js';
 import type { IRational } from '../../lib/types.js';
 /**
  * Timestamp and time base utilities.
@@ -36,6 +36,7 @@ export declare class TimestampUtils {
      * Direct mapping to av_ts2str()
      *
      * @param ts - Timestamp value (bigint or number), or null
+     *
      * @returns String representation
      *
      * @example
@@ -56,7 +57,9 @@ export declare class TimestampUtils {
      * Direct mapping to av_ts2timestr()
      *
      * @param ts - Timestamp value
+     *
      * @param timeBase - Time base for conversion
+     *
      * @returns Time string in seconds with decimal places
      *
      * @example
@@ -78,9 +81,13 @@ export declare class TimestampUtils {
      * Direct mapping to av_compare_ts()
      *
      * @param tsA - First timestamp
+     *
      * @param tbA - Time base of first timestamp
+     *
      * @param tsB - Second timestamp
+     *
      * @param tbB - Time base of second timestamp
+     *
      * @returns -1 if tsA < tsB, 0 if equal, 1 if tsA > tsB
      *
      * @example
@@ -107,8 +114,11 @@ export declare class TimestampUtils {
      * Direct mapping to av_rescale_q()
      *
      * @param a - Timestamp to rescale
+     *
      * @param bq - Source time base
+     *
      * @param cq - Destination time base
+     *
      * @returns Rescaled timestamp
      *
      * @example
@@ -133,9 +143,13 @@ export declare class TimestampUtils {
      * Direct mapping to av_rescale_rnd()
      *
      * @param a - Value to rescale
+     *
      * @param b - Multiplier
+     *
      * @param c - Divisor
+     *
      * @param rnd - Rounding mode
+     *
      * @returns Rescaled value: a * b / c
      *
      * @example

package/dist/api/utilities/timestamp.js

@@ -36,6 +36,7 @@ export class TimestampUtils {
      * Direct mapping to av_ts2str()
      *
      * @param ts - Timestamp value (bigint or number), or null
+     *
      * @returns String representation
      *
      * @example
@@ -58,7 +59,9 @@ export class TimestampUtils {
      * Direct mapping to av_ts2timestr()
      *
      * @param ts - Timestamp value
+     *
      * @param timeBase - Time base for conversion
+     *
      * @returns Time string in seconds with decimal places
      *
      * @example
@@ -82,9 +85,13 @@ export class TimestampUtils {
      * Direct mapping to av_compare_ts()
      *
      * @param tsA - First timestamp
+     *
      * @param tbA - Time base of first timestamp
+     *
      * @param tsB - Second timestamp
+     *
      * @param tbB - Time base of second timestamp
+     *
      * @returns -1 if tsA < tsB, 0 if equal, 1 if tsA > tsB
      *
      * @example
@@ -113,8 +120,11 @@ export class TimestampUtils {
      * Direct mapping to av_rescale_q()
      *
      * @param a - Timestamp to rescale
+     *
      * @param bq - Source time base
+     *
      * @param cq - Destination time base
+     *
      * @returns Rescaled timestamp
      *
      * @example
@@ -141,9 +151,13 @@ export class TimestampUtils {
      * Direct mapping to av_rescale_rnd()
      *
      * @param a - Value to rescale
+     *
      * @param b - Multiplier
+     *
      * @param c - Divisor
+     *
      * @param rnd - Rounding mode
+     *
      * @returns Rescaled value: a * b / c
      *
      * @example

package/dist/api/utilities/timestamp.js.map

@@ -1 +1 @@
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