@mediaproc/audio 1.0.0

package/README.md

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+# Audio Plugin
+
+Professional audio processing plugin for MediaProc CLI. Convert, extract, trim, merge, and normalize audio files with FFmpeg.
+
+## Features
+
+- 🔄 **Convert** - Convert between audio formats (MP3, AAC, WAV, FLAC, OGG, Opus)
+- 🎵 **Extract** - Extract audio tracks from video files
+- ✂️ **Trim** - Cut audio to specific time ranges with fade effects
+- 🔗 **Merge** - Concatenate multiple audio files with crossfade
+- 📊 **Normalize** - EBU R128 loudness normalization for consistent levels
+
+## Installation
+
+```bash
+npm install -g @mediaproc/audio
+```
+
+Or via MediaProc CLI:
+
+```bash
+mediaproc add audio
+```
+
+## Requirements
+
+- **FFmpeg** 4.0+ with audio codecs support
+- **Node.js** 18+
+
+Check if FFmpeg is installed:
+
+```bash
+ffmpeg -version
+```
+
+## Commands
+
+### convert - Format Conversion
+
+Convert audio files between different formats with quality control.
+
+**Supported Formats:** MP3, AAC, WAV, FLAC, OGG, Opus, M4A
+
+```bash
+# Basic conversion
+mediaproc audio convert input.wav -f mp3
+
+# High-quality conversion
+mediaproc audio convert input.flac -f mp3 -b 320k
+
+# Lossless conversion
+mediaproc audio convert input.wav -f flac -q lossless
+
+# Change sample rate and channels
+mediaproc audio convert input.mp3 -f wav -s 48000 -c 2
+
+# Batch convert folder
+mediaproc audio convert audio-files/ -f aac -o output/
+```
+
+**Options:**
+
+- `-o, --output <path>` - Output file/directory
+- `-f, --format <format>` - Output format (mp3, aac, wav, flac, ogg, opus, m4a)
+- `-b, --bitrate <bitrate>` - Audio bitrate (e.g., 128k, 192k, 320k)
+- `-q, --quality <quality>` - Quality preset: low, medium, high, lossless
+- `-s, --sample-rate <rate>` - Sample rate in Hz (44100, 48000)
+- `-c, --channels <channels>` - Channels: 1 (mono), 2 (stereo)
+
+---
+
+### extract - Audio from Video
+
+Extract audio tracks from video files to standalone audio files.
+
+**Supported Video Formats:** MP4, MKV, AVI, MOV, WebM, FLV
+
+```bash
+# Extract as MP3 (default)
+mediaproc audio extract video.mp4
+
+# Extract as high-quality AAC
+mediaproc audio extract video.mp4 -f aac -b 256k
+
+# Extract as lossless FLAC
+mediaproc audio extract video.mkv -f flac -q lossless
+
+# Extract as mono for voice
+mediaproc audio extract video.mp4 -f mp3 --channels 1
+
+# Batch extract from folder
+mediaproc audio extract videos/ -f mp3 -o audio-tracks/
+```
+
+**Options:**
+
+- `-o, --output <path>` - Output file/directory
+- `-f, --format <format>` - Output format (mp3, aac, wav, flac, opus, ogg)
+- `-b, --bitrate <bitrate>` - Audio bitrate
+- `-q, --quality <quality>` - Quality preset
+- `--sample-rate <rate>` - Sample rate
+- `--channels <channels>` - Number of channels
+
+---
+
+### trim - Cut Audio
+
+Trim audio files to specific time ranges with optional fade effects.
+
+```bash
+# Trim by time range
+mediaproc audio trim audio.mp3 --start 00:01:00 --end 00:02:00
+
+# Extract first 30 seconds
+mediaproc audio trim audio.mp3 --duration 30
+
+# Trim from 30s for 60s
+mediaproc audio trim audio.mp3 -s 30 -d 60
+
+# Add fade effects
+mediaproc audio trim audio.mp3 -s 60 -d 120 --fade-in 2 --fade-out 3
+
+# Fast mode (stream copy, no re-encoding)
+mediaproc audio trim audio.mp3 --start 30 --duration 60 --fast
+
+# Batch trim all files
+mediaproc audio trim folder/ -s 10 -d 30 -o output/
+```
+
+**Options:**
+
+- `-o, --output <path>` - Output file/directory
+- `-s, --start <time>` - Start time (HH:MM:SS or seconds)
+- `-e, --end <time>` - End time (HH:MM:SS or seconds)
+- `-d, --duration <time>` - Duration from start
+- `--fade-in <seconds>` - Fade-in duration
+- `--fade-out <seconds>` - Fade-out duration
+- `--fast` - Fast mode (stream copy)
+
+---
+
+### merge - Concatenate Audio
+
+Join multiple audio files into one continuous file.
+
+```bash
+# Basic merge
+mediaproc audio merge audio1.mp3 audio2.mp3 audio3.mp3
+
+# Merge with custom output
+mediaproc audio merge part*.mp3 -o complete.mp3
+
+# Merge with crossfade
+mediaproc audio merge song1.mp3 song2.mp3 --crossfade 2
+
+# Merge with normalization
+mediaproc audio merge *.wav --normalize -o normalized.wav
+
+# Merge to different format
+mediaproc audio merge *.wav -o output.flac --format flac
+```
+
+**Options:**
+
+- `-o, --output <path>` - Output file (default: merged.mp3)
+- `--format <format>` - Output format
+- `--bitrate <bitrate>` - Output bitrate
+- `--crossfade <seconds>` - Crossfade duration between files
+- `--normalize` - Normalize audio levels before merging
+
+---
+
+### normalize - Loudness Normalization
+
+Normalize audio levels using EBU R128 loudness standard for consistent volume.
+
+```bash
+# Normalize to broadcast standard (-16 LUFS)
+mediaproc audio normalize audio.mp3
+
+# Normalize to streaming standard (-23 LUFS)
+mediaproc audio normalize audio.mp3 -t -23
+
+# Custom target loudness
+mediaproc audio normalize audio.mp3 -t -16 -l -1.0
+
+# Simple peak normalization
+mediaproc audio normalize audio.mp3 -m peak
+
+# Batch normalize folder
+mediaproc audio normalize folder/ -o output/
+```
+
+**Options:**
+
+- `-o, --output <path>` - Output file/directory
+- `-t, --target <lufs>` - Target loudness in LUFS (default: -16)
+- `-l, --max-level <db>` - Maximum true peak in dB (default: -1.5)
+- `-m, --method <method>` - Method: loudnorm (EBU R128), peak
+- `--format <format>` - Output format
+
+**LUFS Standards:**
+
+- **-16 LUFS** - Broadcast standard (TV, radio)
+- **-23 LUFS** - Streaming platforms (Spotify, YouTube)
+- **-14 LUFS** - Apple Music, Tidal
+
+---
+
+## Format Support
+
+| Format | Extension      | Type     | Codec      | Best For                   |
+| ------ | -------------- | -------- | ---------- | -------------------------- |
+| MP3    | `.mp3`         | Lossy    | libmp3lame | Universal compatibility    |
+| AAC    | `.aac`, `.m4a` | Lossy    | aac        | Modern devices, streaming  |
+| WAV    | `.wav`         | Lossless | pcm_s16le  | Professional editing       |
+| FLAC   | `.flac`        | Lossless | flac       | Archival, lossless quality |
+| OGG    | `.ogg`         | Lossy    | libvorbis  | Open-source projects       |
+| Opus   | `.opus`        | Lossy    | libopus    | Voice, low bandwidth       |
+
+## Quality Guidelines
+
+### Bitrate Recommendations
+
+| Quality | MP3  | AAC  | Opus | Use Case                  |
+| ------- | ---- | ---- | ---- | ------------------------- |
+| Low     | 96k  | 64k  | 48k  | Voice, podcasts           |
+| Medium  | 192k | 128k | 96k  | General music             |
+| High    | 320k | 256k | 128k | High-quality distribution |
+
+### Sample Rates
+
+- **44100 Hz** - CD quality (default for most content)
+- **48000 Hz** - Professional audio, video production
+- **96000 Hz** - High-resolution audio
+
+## Common Workflows
+
+### Podcast Production
+
+```bash
+# Extract audio from video recording
+mediaproc audio extract recording.mp4 -f mp3 -b 128k --channels 1
+
+# Normalize levels
+mediaproc audio normalize recording-audio.mp3 -t -16
+
+# Trim intro/outro
+mediaproc audio trim recording-audio-normalized.mp3 -s 10 --duration 3600
+```
+
+### Music Production
+
+```bash
+# Convert to lossless for editing
+mediaproc audio convert track.mp3 -f wav -q lossless
+
+# Merge multiple takes
+mediaproc audio merge take1.wav take2.wav take3.wav --normalize
+
+# Export final master
+mediaproc audio convert merged.wav -f flac -q lossless
+```
+
+### Video Production
+
+```bash
+# Extract audio from video
+mediaproc audio extract video.mp4 -f wav
+
+# Clean up audio
+mediaproc audio normalize video-audio.wav -t -16
+
+# Trim to match video
+mediaproc audio trim video-audio-normalized.wav -s 0 -e 00:05:30
+```
+
+## Advanced Examples
+
+### Batch Processing
+
+```bash
+# Convert all WAV files to MP3
+for file in *.wav; do
+  mediaproc audio convert "$file" -f mp3 -b 320k
+done
+
+# Or use folder processing
+mediaproc audio convert wav-files/ -f mp3 -b 320k -o mp3-files/
+```
+
+### Creating Audio Montage
+
+```bash
+# Merge with crossfades and normalization
+mediaproc audio merge \
+  intro.mp3 \
+  main-content.mp3 \
+  outro.mp3 \
+  --crossfade 2 \
+  --normalize \
+  -o final-montage.mp3
+```
+
+### Format Migration
+
+```bash
+# Migrate entire music library from MP3 to FLAC
+mediaproc audio convert music-library/ \
+  -f flac \
+  -q lossless \
+  -o flac-library/
+```
+
+## Troubleshooting
+
+### FFmpeg Not Found
+
+```bash
+# macOS
+brew install ffmpeg
+
+# Ubuntu/Debian
+sudo apt install ffmpeg
+
+# Windows (via Chocolatey)
+choco install ffmpeg
+```
+
+### Audio Quality Issues
+
+- Use lossless formats (WAV, FLAC) for intermediate processing
+- Avoid multiple lossy conversions (MP3 → AAC → OGG)
+- Use higher bitrates (320k) for final distribution
+- Enable normalization for consistent loudness
+
+### Merge Compatibility Issues
+
+Different formats may require re-encoding:
+
+```bash
+# Force re-encoding with specific format
+mediaproc audio merge *.* --format mp3 --bitrate 192k
+```
+
+## Performance Tips
+
+- Use `--fast` mode for large trim operations
+- Enable batch processing for multiple files
+- Consider hardware acceleration for video extraction
+- Use appropriate quality settings (avoid overkill)
+
+## License
+
+MIT
+
+## Contributing
+
+Issues and PRs welcome at [mediaproc-cli](https://github.com/0xshariq/mediaproc-cli)

package/bin/cli.js

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+#!/usr/bin/env node
+import('../dist/cli.js').catch((error) => {
+  console.error('Fatal error:', error);
+  process.exit(1);
+});

package/dist/cli.d.ts

@@ -0,0 +1,3 @@
+#!/usr/bin/env node
+export {};
+//# sourceMappingURL=cli.d.ts.map

package/dist/cli.d.ts.map

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package/dist/cli.js

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+#!/usr/bin/env node
+import { Command } from 'commander';
+import { register } from './register.js';
+const program = new Command();
+program
+    .name('mediaproc-audio')
+    .description(`
+🎵 Audio Processing Plugin v1.0.0
+
+Professional audio processing powered by FFmpeg/FFprobe. Convert, normalize, trim, merge, and extract audio with 5 powerful commands.
+
+📦 Commands Overview:
+
+  🔄 convert [input]
+     Convert audio between formats with quality control
+     • Formats: MP3, AAC, WAV, FLAC, OGG, Opus, M4A, WMA
+     • Quality presets: low (96k), medium (192k), high (320k), lossless
+     • Custom bitrate, sample rate, and channel control
+     • Example: mediaproc-audio convert song.wav -f mp3 -b 320k
+
+  🎵 extract [input]
+     Extract audio tracks from video files
+     • Input: MP4, MKV, AVI, MOV, WebM, FLV, WMV, M4V
+     • Output: MP3, AAC, WAV, FLAC, Opus, OGG
+     • Quality and bitrate control
+     • Example: mediaproc-audio extract video.mp4 -f flac -q lossless
+
+  📊 normalize [input]
+     Normalize audio levels to consistent loudness
+     • EBU R128 loudness normalization (loudnorm filter)
+     • Peak normalization method
+     • Target LUFS: -16 (broadcast), -23 (streaming), -14 (podcasts)
+     • True peak limiting to prevent clipping
+     • Example: mediaproc-audio normalize podcast.mp3 -t -16 -l -1.5
+
+  ✂️  trim [input]
+     Cut audio segments with precise timing
+     • Time-based: HH:MM:SS or seconds (e.g., 00:01:30 or 90)
+     • Duration-based: extract specific length
+     • Optional fade-in/fade-out effects (0.1-10 seconds)
+     • Fast mode: stream copy without re-encoding
+     • Example: mediaproc-audio trim song.mp3 -s 30 -d 60 --fade-in 2
+
+  🔗 merge [inputs...]
+     Concatenate multiple audio files into one
+     • Seamless joining of audio tracks
+     • Optional crossfade between files (0-10 seconds)
+     • Automatic format normalization
+     • Audio level normalization option
+     • Example: mediaproc-audio merge part1.mp3 part2.mp3 -o complete.mp3 --crossfade 2
+
+🎶 Format Support:
+  Lossy Formats:    MP3, AAC, OGG Vorbis, Opus, M4A (AAC), WMA
+  Lossless Formats: WAV (PCM), FLAC (Free Lossless Audio Codec)
+  Video Sources:    MP4, MKV, AVI, MOV, WebM, FLV, WMV, M4V
+
+📡 Audio Quality Guide:
+  96k  (low)      - Voice recordings, podcasts, audiobooks
+  128k (medium-)  - Acceptable music quality, streaming
+  192k (medium)   - Good music quality, general use
+  256k (high-)    - Very good quality, near-transparent
+  320k (high)     - Maximum MP3 quality, transparent to most listeners
+  FLAC/WAV        - Lossless, archival quality, professional mastering
+
+📊 Loudness Normalization Standards:
+  -16 LUFS - EBU R128 broadcast standard (TV, Radio, Streaming)
+  -23 LUFS - ATSC A/85 loudness target (Spotify, YouTube, Apple Music)
+  -14 LUFS - Podcast and audiobook standard
+  -9 LUFS  - Mastering reference (loud commercial sound)
+
+🔧 Common Workflows:
+  # Convert WAV to high-quality MP3
+  mediaproc-audio convert master.wav -f mp3 -b 320k -s 48000
+
+  # Extract lossless audio from video
+  mediaproc-audio extract concert.mp4 -f flac -q lossless
+
+  # Normalize for streaming platforms
+  mediaproc-audio normalize song.mp3 -t -23 -m loudnorm
+
+  # Trim with fade effects
+  mediaproc-audio trim interview.mp3 -s 00:05:30 -e 00:15:45 --fade-in 1 --fade-out 2
+
+  # Merge album tracks with crossfade
+  mediaproc-audio merge track*.mp3 -o album.mp3 --crossfade 3 --normalize
+
+  # Batch convert folder to Opus
+  for file in *.wav; do mediaproc-audio convert "$file" -f opus -b 192k; done
+
+🚀 Quick Start:
+  mediaproc-audio <command> [input] [options]
+  
+📚 Detailed Help:
+  Use 'mediaproc-audio <command> --help' for comprehensive documentation on each command.
+  `)
+    .version('1.0.0');
+register(program);
+program.parse(process.argv);
+if (!process.argv.slice(2).length) {
+    program.outputHelp();
+}
+//# sourceMappingURL=cli.js.map

package/dist/cli.js.map

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package/dist/commands/convert.d.ts

@@ -0,0 +1,3 @@
+import type { Command } from 'commander';
+export declare function convertCommand(audioCmd: Command): void;
+//# sourceMappingURL=convert.d.ts.map

package/dist/commands/convert.d.ts.map

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package/dist/commands/convert.js

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+import chalk from 'chalk';
+import { stat } from 'fs/promises';
+import { runFFmpeg, getAudioMetadata, checkFFmpeg, formatFileSize, formatDuration, } from '../utils/ffmpeg.js';
+import { parseInputPaths, resolveOutputPaths, validateOutputPath } from '../utils/pathValidator.js';
+import { createStandardHelp } from '../utils/helpFormatter.js';
+import ora from 'ora';
+export function convertCommand(audioCmd) {
+    audioCmd
+        .command('convert [input]')
+        .description('Convert audio files between different formats')
+        .option('-o, --output <path>', 'Output file or directory path')
+        .option('-f, --format <format>', 'Output format: mp3, aac, wav, flac, ogg, opus, m4a', 'mp3')
+        .option('-b, --bitrate <bitrate>', 'Audio bitrate (e.g., 128k, 192k, 320k)', '192k')
+        .option('-s, --sample-rate <rate>', 'Sample rate in Hz (e.g., 44100, 48000)', parseInt)
+        .option('-c, --channels <channels>', 'Number of channels: 1 (mono), 2 (stereo)', parseInt)
+        .option('-q, --quality <quality>', 'Quality preset: low, medium, high, lossless', 'medium')
+        .option('--codec <codec>', 'Audio codec override (libmp3lame, aac, flac, libvorbis, libopus)')
+        .option('--dry-run', 'Preview command without executing')
+        .option('-v, --verbose', 'Show detailed FFmpeg output')
+        .option('-h, --help', 'Display help for convert command')
+        .action(async (input, options) => {
+        if (options.help || !input) {
+            createStandardHelp({
+                commandName: 'convert',
+                emoji: '🔄',
+                description: 'Convert audio files between formats (MP3, AAC, WAV, FLAC, OGG, Opus). Supports quality presets, bitrate control, and sample rate adjustment.',
+                usage: [
+                    'convert <input> [options]',
+                    'convert audio.wav -f mp3',
+                    'convert audio-files/ -f aac -o output/'
+                ],
+                options: [
+                    { flag: '-o, --output <path>', description: 'Output file/directory path (default: <input>-converted.<ext>)' },
+                    { flag: '-f, --format <format>', description: 'Output format: mp3, aac, wav, flac, ogg, opus, m4a (default: mp3)' },
+                    { flag: '-b, --bitrate <bitrate>', description: 'Audio bitrate: 128k, 192k, 256k, 320k (default: 192k)' },
+                    { flag: '-s, --sample-rate <rate>', description: 'Sample rate: 44100 (CD quality), 48000 (studio), 96000 (Hi-Res)' },
+                    { flag: '-c, --channels <channels>', description: 'Audio channels: 1 (mono), 2 (stereo)' },
+                    { flag: '-q, --quality <quality>', description: 'Quality preset: low (96k), medium (192k), high (320k), lossless' },
+                    { flag: '--codec <codec>', description: 'Codec override: libmp3lame, aac, flac, libvorbis, libopus' },
+                    { flag: '--dry-run', description: 'Preview FFmpeg command without executing' },
+                    { flag: '-v, --verbose', description: 'Show detailed FFmpeg output and progress' }
+                ],
+                examples: [
+                    { command: 'convert audio.wav -f mp3', description: 'Convert WAV to MP3' },
+                    { command: 'convert audio.mp3 -f flac -q lossless', description: 'Convert to lossless FLAC' },
+                    { command: 'convert audio.flac -f mp3 -b 320k', description: 'Convert FLAC to high-quality MP3' },
+                    { command: 'convert audio.mp3 -f aac -b 128k', description: 'Convert to AAC with specific bitrate' },
+                    { command: 'convert audio.wav -f mp3 -s 48000', description: 'Convert with specific sample rate' },
+                    { command: 'convert folder/ -f mp3 -o output/', description: 'Batch convert all audio files in folder' }
+                ],
+            });
+            return;
+        }
+        try {
+            // Check FFmpeg availability
+            const ffmpegAvailable = await checkFFmpeg();
+            if (!ffmpegAvailable) {
+                console.error(chalk.red('\n✗ FFmpeg not found. Please install FFmpeg first.'));
+                process.exit(1);
+            }
+            // Parse and validate input/output paths
+            const inputPaths = parseInputPaths(input, {
+                allowedExtensions: ['.mp3', '.wav', '.flac', '.aac', '.ogg', '.opus', '.m4a', '.wma']
+            });
+            const outputDir = validateOutputPath(options.output);
+            const outputPathsMap = resolveOutputPaths(inputPaths, outputDir, {
+                suffix: '-converted',
+                newExtension: `.${options.format}`
+            });
+            const outputPaths = Array.from(outputPathsMap.values());
+            // Quality presets
+            const qualityMap = {
+                low: '96k',
+                medium: '192k',
+                high: '320k',
+                lossless: 'lossless'
+            };
+            const targetBitrate = qualityMap[options.quality] || options.bitrate;
+            // Process each file
+            for (let i = 0; i < inputPaths.length; i++) {
+                const inputFile = inputPaths[i];
+                const outputFile = outputPaths[i];
+                console.log(chalk.blue(`\n🔄 Converting: ${inputFile}`));
+                // Get metadata
+                const metadata = await getAudioMetadata(inputFile);
+                const inputStat = await stat(inputFile);
+                console.log(chalk.dim(`Duration: ${formatDuration(metadata.duration)} • ` +
+                    `Sample Rate: ${metadata.sampleRate} Hz • ` +
+                    `Channels: ${metadata.channels}`));
+                // Build FFmpeg args
+                const args = ['-i', inputFile, '-y'];
+                // Codec selection
+                const codecMap = {
+                    mp3: 'libmp3lame',
+                    aac: 'aac',
+                    m4a: 'aac',
+                    flac: 'flac',
+                    wav: 'pcm_s16le',
+                    ogg: 'libvorbis',
+                    opus: 'libopus',
+                };
+                const codec = options.codec || codecMap[options.format];
+                if (codec) {
+                    args.push('-c:a', codec);
+                }
+                // Bitrate
+                if (targetBitrate !== 'lossless') {
+                    args.push('-b:a', targetBitrate);
+                }
+                // Sample rate
+                if (options.sampleRate) {
+                    args.push('-ar', options.sampleRate.toString());
+                }
+                // Channels
+                if (options.channels) {
+                    args.push('-ac', options.channels.toString());
+                }
+                args.push(outputFile);
+                // Dry run
+                if (options.dryRun) {
+                    console.log(chalk.yellow('\n[DRY RUN] Would execute:'));
+                    console.log(chalk.dim(`ffmpeg ${args.join(' ')}`));
+                    continue;
+                }
+                // Execute conversion
+                const spinner = ora('Converting...').start();
+                try {
+                    await runFFmpeg(args, options.verbose);
+                    const outputStat = await stat(outputFile);
+                    spinner.succeed(chalk.green('Conversion complete'));
+                    console.log(chalk.green(`✓ Output: ${outputFile}`));
+                    console.log(chalk.dim(`Size: ${formatFileSize(inputStat.size)} → ${formatFileSize(outputStat.size)}`));
+                }
+                catch (error) {
+                    spinner.fail(chalk.red('Conversion failed'));
+                    throw error;
+                }
+            }
+            if (inputPaths.length > 1) {
+                console.log(chalk.green(`\n✓ Converted ${inputPaths.length} files successfully`));
+            }
+        }
+        catch (error) {
+            console.error(chalk.red(`\n✗ Error: ${error.message}`));
+            process.exit(1);
+        }
+    });
+}
+//# sourceMappingURL=convert.js.map

package/dist/commands/convert.js.map

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package/dist/commands/extract.d.ts

@@ -0,0 +1,3 @@
+import type { Command } from 'commander';
+export declare function extractCommand(audioCmd: Command): void;
+//# sourceMappingURL=extract.d.ts.map

package/dist/commands/extract.d.ts.map

@@ -0,0 +1 @@
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