# Peasy Audio — Full Catalog > Audio tools made easy. Peasy Audio provides free browser-based tools at https://peasyaudio.com/. All processing happens client-side — files never leave your device. ## Tools (15) ### BPM Calculator Calculate beat durations, bar lengths, and delay times from any BPM. Shows note values from whole notes to 32nd notes with dotted and triplet variants. - URL: https://peasyaudio.com/audio/audio-bpm/ - Category: Audio Tools - Processing: client ### Frequency Calculator Convert between note names, frequencies (Hz), MIDI numbers, and wavelengths. Supports custom concert pitch from A4=432 to 443 Hz. - URL: https://peasyaudio.com/audio/audio-freq/ - Category: Audio Tools - Processing: client ### Delay Time Calculator Generate a complete delay time table from any BPM. Shows ms values for every note division — normal, dotted, and triplet. - URL: https://peasyaudio.com/audio/audio-delay/ - Category: Audio Tools - Processing: client ### dB Converter Convert between decibels and linear amplitude or power values. Shows voltage and power ratios with common reference points. - URL: https://peasyaudio.com/audio/audio-db/ - Category: Audio Tools - Processing: client ### Audio File Size Calculator Calculate audio file sizes from duration, sample rate, bit depth, and channels. Compare across WAV, FLAC, MP3, AAC, and Opus. - URL: https://peasyaudio.com/audio/audio-filesize/ - Category: Audio Tools - Processing: client ### Note Frequency Chart Complete chromatic scale reference chart with frequencies, MIDI numbers, and wavelengths from C0 to B8. - URL: https://peasyaudio.com/audio/audio-note-chart/ - Category: Audio Tools - Processing: client ### Decibel Reference Guide Reference guide of common sound levels in dB SPL with real-world examples and OSHA exposure limits. - URL: https://peasyaudio.com/audio/audio-db-reference/ - Category: Audio Tools - Processing: client ### Latency Calculator Calculate audio latency from sample rate and buffer size. See one-way and round-trip latency with performance ratings. - URL: https://peasyaudio.com/audio/audio-latency/ - Category: Audio Tools - Processing: client ### Cents Calculator Calculate pitch deviation in cents between two frequencies. 100 cents = 1 semitone. Useful for tuning and microtonal work. - URL: https://peasyaudio.com/audio/audio-cents/ - Category: Audio Tools - Processing: client ### Loudness Standards Guide Loudness normalization targets for Spotify, Apple Music, YouTube, broadcast, and podcast. LUFS, true peak, and loudness range. - URL: https://peasyaudio.com/audio/audio-loudness/ - Category: Audio Tools - Processing: client ### EQ Frequency Reference Frequency band guide for mixing and mastering Interactive EQ frequency reference with band descriptions, instrument ranges, and genre-specific presets. Identify problem frequencies and learn where to cut or boost. - URL: https://peasyaudio.com/audio/audio-eq-reference/ - Category: Audio Tools - Processing: client - Steps: 1. 1 2. . 3. 4. 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A 169. W ### Sample Rate Converter Compare sample rates and estimate file sizes Convert between audio sample rates and calculate file sizes, Nyquist frequencies, and quality differences. Compare CD, DVD, hi-res, and professional rates. - URL: https://peasyaudio.com/audio/audio-sample-rate-converter/ - Category: Audio Tools - Processing: client - Steps: 1. 1 2. . 3. 4. E 5. n 6. t 7. e 8. r 9. 10. a 11. 12. s 13. a 14. m 15. p 16. l 17. e 18. 19. r 20. a 21. t 22. e 23. 24. o 25. r 26. 27. s 28. e 29. l 30. e 31. c 32. t 33. 34. a 35. 36. s 37. t 38. a 39. n 40. d 41. a 42. r 43. d 44. 45. 2 46. . 47. 48. O 49. p 50. t 51. i 52. o 53. n 54. a 55. l 56. l 57. y 58. 59. s 60. e 61. t 62. 63. b 64. i 65. t 66. 67. d 68. e 69. p 70. t 71. h 72. 73. a 74. n 75. d 76. 77. c 78. h 79. a 80. n 81. n 82. e 83. l 84. s 85. 86. 3 87. . 88. 89. R 90. e 91. v 92. i 93. e 94. w 95. 96. N 97. y 98. q 99. u 100. i 101. s 102. t 103. 104. f 105. r 106. e 107. q 108. u 109. e 110. n 111. c 112. y 113. 114. a 115. n 116. d 117. 118. q 119. u 120. a 121. l 122. i 123. t 124. y 125. 126. t 127. i 128. e 129. r 130. 131. 4 132. . 133. 134. C 135. o 136. m 137. p 138. a 139. r 140. e 141. 142. f 143. i 144. l 145. e 146. 147. s 148. i 149. z 150. e 151. s 152. 153. a 154. c 155. r 156. o 157. s 158. s 159. 160. f 161. o 162. r 163. m 164. a 165. t 166. s ### Bit Depth Analyzer Dynamic range and noise floor by bit depth Analyze audio bit depth to understand dynamic range, signal-to-noise ratio, and quantization noise. Compare 8-bit through 64-bit float with dither recommendations. - URL: https://peasyaudio.com/audio/audio-bit-depth-analyzer/ - Category: Audio Tools - Processing: client - Steps: 1. 1 2. . 3. 4. E 5. n 6. t 7. e 8. r 9. 10. a 11. 12. b 13. i 14. t 15. 16. d 17. e 18. p 19. t 20. h 21. 22. o 23. r 24. 25. s 26. e 27. l 28. e 29. c 30. t 31. 32. a 33. 34. s 35. t 36. a 37. n 38. d 39. a 40. r 41. d 42. 43. 2 44. . 45. 46. R 47. e 48. v 49. i 50. e 51. w 52. 53. d 54. y 55. n 56. a 57. m 58. i 59. c 60. 61. r 62. a 63. n 64. g 65. e 66. , 67. 68. S 69. N 70. R 71. , 72. 73. a 74. n 75. d 76. 77. n 78. o 79. i 80. s 81. e 82. 83. f 84. l 85. o 86. o 87. r 88. 89. 3 90. . 91. 92. C 93. o 94. m 95. p 96. a 97. r 98. e 99. 100. m 101. u 102. l 103. t 104. i 105. p 106. l 107. e 108. 109. b 110. i 111. t 112. 113. d 114. e 115. p 116. t 117. h 118. s 119. 120. s 121. i 122. d 123. e 124. 125. b 126. y 127. 128. s 129. i 130. d 131. e 132. 133. 4 134. . 135. 136. G 137. e 138. t 139. 140. d 141. i 142. t 143. h 144. e 145. r 146. i 147. n 148. g 149. 150. r 151. e 152. c 153. o 154. m 155. m 156. e 157. n 158. d 159. a 160. t 161. i 162. o 163. n 164. s 165. 166. f 167. o 168. r 169. 170. c 171. o 172. n 173. v 174. e 175. r 176. s 177. i 178. o 179. n ### Chord Reference Look up chord voicings, intervals, and notes for any chord. 18 chord types including major, minor, 7th, diminished, augmented, and suspended. Piano keyboard visualization. - URL: https://peasyaudio.com/audio/chord-reference/ - Category: Audio Tools - Processing: client ### Key & Scale Reference Look up musical keys, scale degrees, modes, and relative major/minor relationships. 14 scales from major and minor to modes, pentatonics, and blues. - URL: https://peasyaudio.com/audio/key-scale-reference/ - Category: Audio Tools - Processing: client ## Guides (51) ### File Format Conversion: A Complete Guide Converting files between formats is a daily task for professionals across every industry. This comprehensive guide covers document, image, audio, and video conversion principles that apply regardless of the specific tool. - URL: https://peasyaudio.com/guides/file-format-conversion-guide/ - Category: How-To - Reading time: 1 min - Words: 224 Key takeaways: - Regardless of the file type, certain principles apply to all conversions. - The most important concept in format conversion is whether data is lost: - Some features don't transfer between formats: - For converting many files at once: - Client-side conversion (in-browser) provides better privacy since files never leave your device. ## Universal Conversion Principles Regardless of the file type, certain principles apply to all conversions. ## Lossy vs Lossless The most important concept in format conversion is whether data is lost: - **Lossless**: FLAC to WAV, PNG to BMP — no quality loss, fully reversible. - **Lossy**: WAV to MP3, PNG to JPEG — some data is permanently discarded. - **Transcoding**: MP3 to AAC — compound quality loss from re-compression. **Golden rule:** Always convert from the highest quality source available. Never convert between lossy formats if you can avoid it. ## Feature Compatibility Some features don't transfer between formats: | Source Feature | May Be Lost In | |---------------|---------------| | Transparency (PNG) | JPEG | | Layers (PSD) | PNG, JPEG | | Formulas (XLSX) | CSV | | Hyperlinks (DOCX) | Plain text | | Animations (GIF) | PNG, JPEG | | Metadata (EXIF) | Varies by tool | ## Batch Conversion For converting many files at once: 1. **Test first**: Convert a small sample and verify quality. 2. **Consistent settings**: Apply the same parameters to all files. 3. **Preserve originals**: Never overwrite source files. 4. **Naming conventions**: Use consistent output file naming. ## Client-Side vs Server-Side Client-side conversion (in-browser) provides better privacy since files never leave your device. Server-side conversion typically handles larger files and more formats but requires uploading your data. ### Audio Format Guide: MP3 vs AAC vs FLAC vs WAV Audio formats balance file size against sound quality. This comparison covers the most common formats and helps you choose the right one for music, podcasts, voice recordings, and professional audio production. - URL: https://peasyaudio.com/guides/audio-format-comparison/ - Category: Comparison - Reading time: 1 min - Words: 253 Key takeaways: - Audio formats fall into two categories: lossy (which discard inaudible data to reduce size) and lossless (which preserve every bit of the original). - MP3 has been the standard compressed audio format since the late 1990s. - AAC (Advanced Audio Coding) achieves better sound quality than MP3 at the same bitrate. - FLAC compresses audio to about 50-70% of the original size with zero quality loss. - WAV files contain uncompressed PCM audio data. ## Lossy vs Lossless Audio formats fall into two categories: lossy (which discard inaudible data to reduce size) and lossless (which preserve every bit of the original). The right choice depends on your use case and audience. ## MP3: Universal Compatibility MP3 has been the standard compressed audio format since the late 1990s. At 128 kbps it's adequate for speech; at 192-320 kbps it's good for music. **Best for:** Maximum compatibility, podcast distribution, email attachments. ## AAC: Better Quality at Same Bitrate AAC (Advanced Audio Coding) achieves better sound quality than MP3 at the same bitrate. It's the default format for Apple Music, YouTube, and most streaming services. **Best for:** Streaming, mobile audio, iTunes/Apple ecosystem. ## FLAC: Lossless Compression FLAC compresses audio to about 50-70% of the original size with zero quality loss. It's the preferred format for audiophiles and music archival. **Best for:** Music archival, audiophile listening, master recordings. ## WAV: Uncompressed Standard WAV files contain uncompressed PCM audio data. They're large but require no decoding, making them ideal for audio editing and professional workflows. **Best for:** Audio editing, professional recording, sound design. ## Format Comparison | Format | Type | Typical Size (per minute) | Quality | |--------|------|--------------------------|--------| | WAV | Uncompressed | ~10 MB | Perfect | | FLAC | Lossless | ~5-7 MB | Perfect | | MP3 320 | Lossy | ~2.5 MB | Excellent | | AAC 256 | Lossy | ~2 MB | Excellent | | MP3 128 | Lossy | ~1 MB | Good | ### Audio Format Comparison: MP3, AAC, FLAC, OGG, and WAV Choosing the right audio format involves trade-offs between file size, audio quality, compatibility, and intended use case. This comparison covers the five most common formats and when to use each one. - URL: https://peasyaudio.com/guides/audio-format-comparison-mp3-aac-flac/ - Category: Comparison - Reading time: 2 min - Words: 341 Key takeaways: - Lossy codecs (MP3, AAC, OGG) permanently discard audio data deemed inaudible by psychoacoustic models. - The oldest widely-supported lossy format (1993). - The successor to MP3, used as the default format by Apple Music, YouTube, and most streaming services. - WAV:** Recording, editing, and mastering (uncompressed working format) - Essential for archiving, mastering, and audiophile playback. ## Format Overview | Format | Type | Typical Bitrate | File Size (4 min song) | Quality | |--------|------|----------------|----------------------|--------| | WAV | Uncompressed | 1,411 kbps | ~42 MB | Reference | | FLAC | Lossless | 800-1,200 kbps | ~25 MB | Identical to source | | AAC | Lossy | 128-256 kbps | ~6-8 MB | Excellent at 256 kbps | | MP3 | Lossy | 128-320 kbps | ~4-10 MB | Good at 320 kbps | | OGG Vorbis | Lossy | 96-320 kbps | ~4-10 MB | Excellent at 192 kbps | ## Lossy vs. Lossless Lossy codecs (MP3, AAC, OGG) permanently discard audio data deemed inaudible by psychoacoustic models. At high bitrates (256+ kbps), the difference from the original is imperceptible to most listeners in normal listening conditions. Lossless codecs (FLAC) compress without discarding data — the decoded output is bit-identical to the original. ## MP3 The oldest widely-supported lossy format (1993). MP3 at 320 kbps is nearly transparent (indistinguishable from the original) for most listeners. At 128 kbps, artifacts become noticeable on complex material — cymbals sound metallic, and reverb tails become grainy. MP3's advantage is universal compatibility: every device and application supports it. ## AAC The successor to MP3, used as the default format by Apple Music, YouTube, and most streaming services. AAC achieves better quality than MP3 at equivalent bitrates — AAC at 128 kbps sounds comparable to MP3 at 160 kbps. The most efficient lossy codec for general use. ## FLAC The standard lossless format. FLAC files are 50-70% of the original WAV size with zero quality loss. Essential for archiving, mastering, and audiophile playback. Supported by most modern devices except Apple's ecosystem (which prefers ALAC, Apple's lossless codec). ## When to Use Each Format - **WAV:** Recording, editing, and … ### Lossless vs Lossy Audio: When Quality Matters and When It Doesn't The lossless vs. lossy debate generates more heat than light. In controlled double-blind tests, most listeners cannot distinguish 256 kbps AAC from lossless — but there are legitimate scenarios where lossless is essential. - URL: https://peasyaudio.com/guides/lossless-vs-lossy-audio-guide/ - Category: Comparison - Reading time: 2 min - Words: 326 Key takeaways: - Lossy audio codecs exploit psychoacoustic masking — the phenomenon where loud sounds make nearby quieter sounds inaudible. - Every lossy encoding cycle degrades quality. - Listening through earbuds on a subway, car speakers, laptop speakers, or Bluetooth headphones — environmental noise and hardware limitations mask any difference between 256 kbps AAC and lossless. - Keep a lossless archive (FLAC) of everything you create or own. - ### Critical Listening Environments In acoustically treated rooms with high-end headphones or monitors, trained listeners can sometimes detect lossy compression artifacts — pre-echo on transients, smearing in the stereo image, and loss of 'air' in high frequencies. ## What Lossy Compression Removes Lossy audio codecs exploit psychoacoustic masking — the phenomenon where loud sounds make nearby quieter sounds inaudible. A 1 kHz tone at 80 dB masks nearby frequencies at 60 dB. The codec identifies and discards these masked components, reducing file size while preserving perceived quality. ## When Lossless Matters ### Production and Mastering Every lossy encoding cycle degrades quality. If you encode to MP3, edit the MP3, then encode again, each generation loses additional data. Production workflows must use lossless (WAV, FLAC, AIFF) as the working format. Only encode to lossy as the final delivery step. ### Archival Archives should preserve maximum quality for future use. Lossy compression is irreversible — you cannot recover discarded data. A FLAC archive can be transcoded to any future format without generational loss. ### Critical Listening Environments In acoustically treated rooms with high-end headphones or monitors, trained listeners can sometimes detect lossy compression artifacts — pre-echo on transients, smearing in the stereo image, and loss of 'air' in high frequencies. These differences are subtle and disappear in normal listening conditions. ## When Lossy Is Sufficient ### Casual Listening Listening through earbuds on a subway, car speakers, laptop speakers, or Bluetooth headphones — environmental noise and hardware limitations mask any difference between 256 kbps AAC and lossless. ### Streaming and Mobile Bandwidth and storage are limited. A 4-minute FLAC track is 25 MB; the same in AAC 256 is 8 MB. Over a library of 1,000 songs, that is 25 GB vs. 8 GB. ### Podcasts and Voice Human speech has a narrower frequency range and less dynamic complexity than music. Podcasts at 128 kbps AAC mono are indistinguishable from lossless. ## The Practical Recommendation Keep a lossless archive (FLAC) of everything you create or own. Distribute in lossy formats … ### Audio Bitrate and Quality: Finding the Right Balance Bitrate determines both file size and audio quality in lossy formats. Understanding the relationship between bitrate, codec efficiency, and perceptual quality helps you choose the optimal setting for each use case. - URL: https://peasyaudio.com/guides/audio-bitrate-quality-guide/ - Category: Best Practice - Reading time: 2 min - Words: 315 Key takeaways: - Bitrate measures the amount of data used to represent one second of audio, expressed in kilobits per second (kbps). - Every second of audio uses the same number of bits. - 'Transparent' means indistinguishable from the lossless source in double-blind testing. - At equivalent bitrates, modern codecs significantly outperform older ones: - Music streaming:** AAC 256 kbps or Opus 128 kbps ## What Bitrate Means Bitrate measures the amount of data used to represent one second of audio, expressed in kilobits per second (kbps). Higher bitrate means more data per second, which generally means higher quality — but codec efficiency determines how much quality a codec extracts from each kilobit. ## Constant vs. Variable Bitrate ### CBR (Constant Bitrate) Every second of audio uses the same number of bits. Simple passages waste bits; complex passages may be under-allocated. Predictable file size but suboptimal quality distribution. ### VBR (Variable Bitrate) The encoder allocates more bits to complex passages (dense orchestral, cymbals) and fewer to simple passages (silence, sustained notes). VBR produces better quality at a given average bitrate but results in slightly unpredictable file sizes. **Recommendation:** Always use VBR for quality-critical applications. ## Quality Thresholds by Format | Format | Transparent Threshold | Good Enough | Minimum Acceptable | |--------|---------------------|-------------|-------------------| | MP3 (LAME V0) | ~245 kbps VBR | 192 kbps | 128 kbps | | AAC (Apple/FDK) | ~192 kbps VBR | 128 kbps | 96 kbps | | OGG Vorbis | ~192 kbps VBR | 128 kbps | 96 kbps | | Opus | ~128 kbps VBR | 96 kbps | 64 kbps | 'Transparent' means indistinguishable from the lossless source in double-blind testing. ## Codec Efficiency Rankings At equivalent bitrates, modern codecs significantly outperform older ones: 1. **Opus** — Best overall efficiency (especially below 128 kbps) 2. **AAC (Apple/FDK)** — Best for music distribution 3. **OGG Vorbis** — Excellent open-source alternative 4. **MP3 (LAME)** — Oldest, least efficient, but most compatible ## Practical Guidelines - **Music streaming:** AAC 256 kbps or Opus 128 kbps - **Podcasts:** AAC 128 kbps mono or Opus 64 kbps mono - **Voice calls:** Opus 32-64 kbps (designed for speech) - **Background music (web):** AAC … ### Audio Format Guide: MP3, AAC, FLAC, WAV, and OGG Explained Understand the differences between lossy and lossless audio formats. Learn when to use each format for music, podcasts, sound effects, and professional audio production. - URL: https://peasyaudio.com/guides/audio-format-guide-mp3-aac-flac-wav-ogg/ - Category: How-To - Reading time: 1 min - Words: 292 Key takeaways: - Lossy codecs (MP3, AAC, OGG Vorbis) remove audio data that psychoacoustic models predict humans cannot perceive. - MP3: Maximum compatibility — email attachments, legacy devices, podcast distribution. - Never convert between lossy formats — each re-encoding compounds quality loss. - Lossless codecs (FLAC, ALAC, WAV) preserve every sample exactly — important for production and archiving but resulting in much larger files. - ## Conversion Best Practices Never convert between lossy formats — each re-encoding compounds quality loss. ## Lossy vs Lossless Audio Lossy codecs (MP3, AAC, OGG Vorbis) remove audio data that psychoacoustic models predict humans cannot perceive. This typically achieves 5-10x compression. Lossless codecs (FLAC, ALAC, WAV) preserve every sample exactly — important for production and archiving but resulting in much larger files. ## Format Comparison | Format | Type | Typical Bitrate | File Size (4 min) | Browser Support | | --- | --- | --- | --- | --- | | MP3 | Lossy | 128-320 kbps | 4-10 MB | Universal | | AAC | Lossy | 96-256 kbps | 3-8 MB | Universal | | OGG Vorbis | Lossy | 96-320 kbps | 3-10 MB | Firefox, Chrome | | Opus | Lossy | 32-256 kbps | 1-8 MB | Modern browsers | | FLAC | Lossless | 800-1200 kbps | 25-40 MB | Chrome, Firefox, Edge | | WAV | Uncompressed | 1411 kbps | 42 MB | Universal | ## When to Use Each **MP3**: Maximum compatibility — email attachments, legacy devices, podcast distribution. Use 192 kbps or higher for music. **AAC**: Superior quality to MP3 at the same bitrate. Default for Apple devices and streaming platforms. Use for web audio where older browser support is needed. **Opus**: The best lossy codec available. Excellent at low bitrates (48-96 kbps) for voice, competitive at higher rates for music. Ideal for WebRTC and streaming. **FLAC**: Lossless archiving and audiophile playback. Compresses WAV to roughly 60% of original size without any quality loss. ## Conversion Best Practices Never convert between lossy formats — each re-encoding compounds quality loss. Always keep lossless originals and create lossy versions as needed. Convert audio files with Peasy's browser-based converter to avoid uploading sensitive audio recordings to external services. ### How to Trim and Edit Audio Files in the Browser Browser-based audio editing lets you trim silence, cut sections, and adjust audio without installing software. Learn how to use the Web Audio API and client-side tools for common audio editing tasks. - URL: https://peasyaudio.com/guides/trim-edit-audio-browser/ - Category: How-To - Reading time: 1 min - Words: 223 Key takeaways: - Modern browsers include the Web Audio API, which provides powerful audio processing capabilities without server uploads. - Trimming removes unwanted audio from the beginning or end of a recording. - The waveform shows amplitude over time. - When exporting edited audio, match the source quality when possible. - This means your audio files never leave your device. ## Client-Side Audio Editing Modern browsers include the Web Audio API, which provides powerful audio processing capabilities without server uploads. This means your audio files never leave your device. ## Common Editing Tasks ### Trimming Trimming removes unwanted audio from the beginning or end of a recording. This is the most frequent editing task for podcasters, voice-over artists, and musicians. **Steps:** 1. Load the audio file into the waveform display. 2. Select the region you want to keep. 3. Delete everything outside the selection. 4. Export the trimmed audio. ### Splitting Split a long recording into multiple files. Useful for breaking a lecture into chapters, separating songs from a live recording, or creating clips from interviews. ### Silence Detection Automatic silence detection identifies pauses in speech or music. You can use this to remove long pauses, split at silence points, or generate timestamps for chapters. ## Waveform Visualization The waveform shows amplitude over time. Louder sections appear taller, silence appears flat. Learning to read waveforms helps you quickly identify: - Where speech starts and ends - Clipping (flat-topped peaks indicate distortion) - Background noise (visible as a thin baseline above zero) ## Export Settings When exporting edited audio, match the source quality when possible. For web distribution, MP3 at 192 kbps or AAC at 128 kbps provides good quality at reasonable file sizes. ### Best Practices for Podcast Audio Quality Clear, consistent audio is the foundation of a professional podcast. This guide covers recording levels, noise reduction, loudness normalization, and export settings that meet podcast platform requirements. - URL: https://peasyaudio.com/guides/podcast-audio-quality-best-practices/ - Category: Best Practice - Reading time: 1 min - Words: 235 Key takeaways: - Podcast platforms have specific audio requirements. - Record with peaks between -12 dB and -6 dB. - Remove background noise like fan hum, air conditioning, or electrical buzz. - ## Recording Best Practices ### Input Levels Record with peaks between -12 dB and -6 dB. ## Audio Quality Standards Podcast platforms have specific audio requirements. Apple Podcasts recommends mono or stereo AAC at 128-256 kbps, while Spotify accepts MP3 at 96-320 kbps. ## Recording Best Practices ### Input Levels Record with peaks between -12 dB and -6 dB. This leaves enough headroom to prevent clipping while capturing a strong signal. Never let peaks hit 0 dB. ### Room Acoustics Record in a quiet room with soft surfaces. Closets full of clothes make surprisingly good recording spaces. Avoid rooms with hard walls, glass, and tile that create echo and reverberation. ## Post-Production ### Noise Reduction Remove background noise like fan hum, air conditioning, or electrical buzz. Capture a few seconds of room tone (silence) at the start of each recording to use as a noise profile. ### Loudness Normalization Normalize to -16 LUFS (stereo) or -19 LUFS (mono) for consistent playback volume across episodes and between different podcasts on the same platform. ### Dynamic Range Compression Light compression (2:1 to 4:1 ratio) evens out volume differences between loud and quiet speech. This makes podcasts easier to listen to in noisy environments like commuting. ## Export Settings | Platform | Format | Bitrate | Sample Rate | |----------|--------|---------|-------------| | Apple Podcasts | AAC | 128-256 kbps | 44.1 kHz | | Spotify | MP3/AAC | 96-320 kbps | 44.1 kHz | | YouTube | AAC | 192+ kbps | 48 kHz | ### How to Convert Between Audio Formats Converting audio files between formats is common when preparing music for different devices, uploading to platforms, or archiving recordings. Learn how to convert without unnecessary quality loss. - URL: https://peasyaudio.com/guides/convert-between-audio-formats/ - Category: How-To - Reading time: 1 min - Words: 238 Key takeaways: - Never convert from one lossy format to another (e.g., MP3 to AAC). - This is the most common and safest conversion direction: - CD-quality audio uses 44.1 kHz. - ## The Golden Rule of Audio Conversion Never convert from one lossy format to another (e. - Always convert from the highest-quality source available. ## The Golden Rule of Audio Conversion Never convert from one lossy format to another (e.g., MP3 to AAC). Each lossy conversion discards more data, compounding quality loss. Always convert from the highest-quality source available. ## Conversion Paths ### Lossless to Lossy (Archival → Distribution) This is the most common and safest conversion direction: - WAV → MP3 (universal compatibility) - FLAC → AAC (streaming platforms) - WAV → OGG (web audio) ### Lossless to Lossless (Format Migration) Converting between lossless formats preserves perfect quality: - WAV → FLAC (reduces size by ~50% with no quality loss) - AIFF → WAV (cross-platform compatibility) ### Lossy to Lossless (Packaging) Converting MP3 to FLAC or WAV doesn't restore lost quality — it just wraps the decoded audio in a lossless container. The file gets larger without improving quality. ## Bitrate Selection Guide | Quality Level | MP3 | AAC | OGG | |--------------|-----|-----|-----| | Acceptable (speech) | 96 kbps | 64 kbps | 80 kbps | | Good (podcasts) | 128 kbps | 96 kbps | 112 kbps | | High (music) | 256 kbps | 192 kbps | 192 kbps | | Transparent | 320 kbps | 256 kbps | 256 kbps | ## Sample Rate Considerations CD-quality audio uses 44.1 kHz. Video audio typically uses 48 kHz. Don't upsample (e.g., 44.1 to 48 kHz) unless required by a specific platform — it increases file size without adding information. ### Troubleshooting Audio Playback Issues Audio files that refuse to play, produce distorted sound, or have sync issues are common frustrations. This guide helps you diagnose and fix the most frequent audio playback problems. - URL: https://peasyaudio.com/guides/troubleshooting-audio-playback/ - Category: Troubleshooting - Reading time: 1 min - Words: 221 Key takeaways: - Audio playback issues typically stem from codec incompatibility, corrupted files, or incorrect metadata. - The most common reason a file won't play is that the player doesn't have the required codec. - Audio that was recorded too loud shows flat-topped waveforms and produces harsh, crackling distortion. - When audio drifts out of sync with video, common causes include: ## Common Playback Problems Audio playback issues typically stem from codec incompatibility, corrupted files, or incorrect metadata. ## File Won't Play ### Codec Not Supported The most common reason a file won't play is that the player doesn't have the required codec. Check the file format and ensure your player supports it. **Quick fix:** Convert to a widely supported format like MP3 or AAC. ### Corrupted File Files can become corrupted during download, transfer, or storage. Signs include playback stopping partway through, clicking sounds, or the file being reported as 0 bytes. **Diagnosis:** Try playing the file in a different application. If it fails everywhere, the file is likely corrupted. ## Distorted Sound ### Clipping Audio that was recorded too loud shows flat-topped waveforms and produces harsh, crackling distortion. Once clipping is baked into a recording, it cannot be fully undone. ### Sample Rate Mismatch Playing a 48 kHz file through a 44.1 kHz output (or vice versa) can cause pitch shifts and artifacts. Ensure your audio interface matches the file's sample rate. ## Audio-Video Sync Issues When audio drifts out of sync with video, common causes include: - Different sample rates between audio and video tracks. - Variable frame rate video (common in screen recordings). - Audio codec decoding latency. **Fix:** Re-encode with matching sample rates and constant frame rate. ### Audio Loudness Standards: LUFS, dBFS, and Normalization Different platforms have different loudness requirements. Understanding LUFS, dBFS, and normalization ensures your audio plays at the right volume on Spotify, Apple Music, YouTube, and broadcast systems. - URL: https://peasyaudio.com/guides/audio-loudness-standards/ - Category: Best Practice - Reading time: 1 min - Words: 238 Key takeaways: - Inconsistent loudness between tracks or episodes creates a poor listening experience. - dBFS measures instantaneous peak levels relative to the maximum digital value. - Adjusts gain so the highest peak reaches a target level. ## Why Loudness Matters Inconsistent loudness between tracks or episodes creates a poor listening experience. Streaming platforms apply their own normalization, which can either turn down or turn up your audio — and turning up reveals noise. ## Key Measurements ### dBFS (Decibels Full Scale) dBFS measures instantaneous peak levels relative to the maximum digital value. 0 dBFS is the absolute maximum — anything higher clips. ### LUFS (Loudness Units Full Scale) LUFS measures perceived loudness over time, accounting for how human ears respond to different frequencies. It's the standard for broadcast and streaming. ## Platform Targets | Platform | Target LUFS | True Peak | |----------|------------|----------| | Spotify | -14 LUFS | -1 dBTP | | Apple Music | -16 LUFS | -1 dBTP | | YouTube | -14 LUFS | -1 dBTP | | Podcast (mono) | -19 LUFS | -1 dBTP | | Broadcast (EU) | -23 LUFS | -1 dBTP | ## Normalization Methods ### Peak Normalization Adjusts gain so the highest peak reaches a target level. Simple but doesn't guarantee consistent perceived loudness. ### Loudness Normalization Adjusts gain so the integrated LUFS matches a target. This produces perceptually consistent volume across different audio content. ## Practical Workflow 1. Mix your audio with peaks around -6 to -3 dBFS. 2. Apply a limiter with a ceiling of -1 dBTP. 3. Measure integrated LUFS. 4. Adjust gain to reach the target LUFS for your platform. ### Container Formats Explained: MP4, MKV, WebM, and MOV Container formats wrap video, audio, and metadata into a single file. Understanding the difference between MP4, MKV, WebM, and MOV helps you choose the right wrapper for your video content. - URL: https://peasyaudio.com/guides/video-container-formats-explained/ - Category: Comparison - Reading time: 1 min - Words: 243 Key takeaways: - A common source of confusion is the difference between codecs and containers. - MP4 is the most widely supported container format. - Matroska (MKV) supports virtually every codec and feature: multiple audio tracks, subtitle tracks, chapters, attachments, and metadata. - WebM is Google's container for VP8, VP9, and AV1 codecs. - QuickTime MOV is Apple's container format. ## Codec vs Container A common source of confusion is the difference between codecs and containers. The codec determines how video is compressed; the container determines how the compressed data is packaged. Think of it like a shipping box (container) holding various items (codec streams, subtitles, metadata). The same item can go in different boxes. ## MP4 (.mp4): Universal Container MP4 is the most widely supported container format. It supports H.264, H.265, AAC, and many other codecs. It's the default for web video, social media, and mobile devices. **Supports:** H.264, H.265, AAC, subtitles (limited). ## MKV (.mkv): The Kitchen Sink Matroska (MKV) supports virtually every codec and feature: multiple audio tracks, subtitle tracks, chapters, attachments, and metadata. It's popular for media libraries. **Supports:** Everything — H.264, H.265, AV1, VP9, FLAC, DTS, chapters. ## WebM (.webm): Web-Optimized WebM is Google's container for VP8, VP9, and AV1 codecs. It's designed for web delivery and is supported natively in all modern browsers. **Supports:** VP8, VP9, AV1, Vorbis, Opus. ## MOV (.mov): Apple's Container QuickTime MOV is Apple's container format. It's functionally similar to MP4 but has better support for ProRes and Apple-specific codecs. **Supports:** H.264, H.265, ProRes, AAC, PCM. ## Choosing the Right Container | Use Case | Recommended Container | |----------|----------------------| | Web delivery | MP4 or WebM | | Media archive | MKV | | Apple ecosystem | MOV | | Social media upload | MP4 | | Professional editing | MOV (ProRes) | ### How to Extract Audio From Video Files Extracting audio tracks from video files is useful for creating podcasts from interviews, music from concerts, and audio-only versions of content. - URL: https://peasyaudio.com/guides/how-to-extract-audio-from-video/ - Category: How-To - Reading time: 1 min - Words: 175 Key takeaways: - Video files contain separate audio and video streams. - When possible, extract without re-encoding. - Sometimes you need the audio in a different format. - Some videos contain multiple audio tracks — different languages, commentary tracks, or music/effects separation. - When converting a series of video lectures or conference talks, batch extraction processes all files with the same settings. ## Why Extract Audio? Video files contain separate audio and video streams. Extracting the audio stream without re-encoding preserves original quality and produces a standalone audio file that's much smaller than the video. ## Lossless Extraction When possible, extract without re-encoding. If the video contains AAC audio, extract it as an .m4a file. If it contains MP3, extract as .mp3. This preserves the original audio quality with zero quality loss. ## Transcoding During Extraction Sometimes you need the audio in a different format. Converting from the video's internal format (e.g., AAC) to MP3 involves one generation of lossy encoding. Use the highest practical bitrate (320 kbps for MP3) to minimize loss. ## Multi-Track Videos Some videos contain multiple audio tracks — different languages, commentary tracks, or music/effects separation. Extract the specific track you need by selecting it by index or language tag. ## Batch Processing When converting a series of video lectures or conference talks, batch extraction processes all files with the same settings. Name output files to match their source videos for easy identification. ### How to Normalize Audio Levels Across Multiple Files Inconsistent volume levels between audio files is distracting. Learn how to normalize audio to a consistent loudness standard. - URL: https://peasyaudio.com/guides/how-to-normalize-audio-levels/ - Category: How-To - Reading time: 1 min - Words: 188 Key takeaways: - Audio files from different sources have different volume levels. - Peak normalization adjusts the loudest moment to a target level. - After normalization, check for true peaks exceeding -1 dBTP. - When normalizing an album or podcast series, process all files together to ensure relative dynamics are preserved. ## Why Normalize? Audio files from different sources have different volume levels. A podcast episode followed by a quiet interview, or a playlist where every song requires volume adjustment, degrades the listening experience. ## Peak vs Loudness Normalization Peak normalization adjusts the loudest moment to a target level. This doesn't account for perceived loudness — a track with one loud spike may still sound quiet overall. Loudness normalization (LUFS-based) adjusts based on perceived loudness, producing more consistent results. ## Target Levels | Platform | Target LUFS | |----------|------------| | Spotify | -14 LUFS | | Apple Music | -16 LUFS | | YouTube | -14 LUFS | | Podcasts | -16 to -18 LUFS | | Broadcast TV | -24 LUFS | ## True Peak Limiting After normalization, check for true peaks exceeding -1 dBTP. Digital clipping causes distortion. Apply a true peak limiter to catch inter-sample peaks that standard meters miss. ## Batch Normalization When normalizing an album or podcast series, process all files together to ensure relative dynamics are preserved. Each file reaches the same perceived loudness without compressing the quiet parts of naturally dynamic content. ### How to Remove Background Noise From Audio Background noise from fans, traffic, and hum can ruin recordings. Learn noise reduction techniques that clean audio without degrading quality. - URL: https://peasyaudio.com/guides/how-to-remove-background-noise/ - Category: How-To - Reading time: 1 min - Words: 172 Key takeaways: - Audio noise falls into categories: broadband (constant hiss from electronics), hum (50/60 Hz from electrical interference), intermittent (clicks, pops), and environmental (traffic, wind, HVAC). - Most noise reduction algorithms work by analyzing a 'noise profile' — a sample of pure noise without speech or music. - The most common technique analyzes the frequency spectrum of noise and subtracts those frequencies from the audio. - Moderate noise reduction that leaves some noise is better than aggressive reduction that creates artifacts. - The best noise reduction happens before recording: use a quality microphone, record in a quiet room, minimize distance to the microphone, use a pop filter, and turn off fans and HVAC during recording. ## Types of Noise Audio noise falls into categories: broadband (constant hiss from electronics), hum (50/60 Hz from electrical interference), intermittent (clicks, pops), and environmental (traffic, wind, HVAC). Each requires different treatment. ## Noise Profiling Most noise reduction algorithms work by analyzing a 'noise profile' — a sample of pure noise without speech or music. The algorithm then subtracts that profile from the entire recording. ## Spectral Subtraction The most common technique analyzes the frequency spectrum of noise and subtracts those frequencies from the audio. Too aggressive subtraction creates 'musical noise' — a warbling artifact that sounds worse than the original noise. ## The 'Less Is More' Principle Moderate noise reduction that leaves some noise is better than aggressive reduction that creates artifacts. Listeners tolerate slight background noise but not metallic warbling or hollow-sounding voices. ## Prevention Over Cure The best noise reduction happens before recording: use a quality microphone, record in a quiet room, minimize distance to the microphone, use a pop filter, and turn off fans and HVAC during recording. ### WAV vs FLAC vs ALAC: Lossless Audio Format Comparison Lossless audio preserves full quality but formats differ in compression, metadata support, and compatibility. Compare WAV, FLAC, and ALAC. - URL: https://peasyaudio.com/guides/wav-vs-flac-vs-alac-comparison/ - Category: Comparison - Reading time: 1 min - Words: 207 Key takeaways: - WAV stores raw PCM audio data with minimal compression. - FLAC compresses audio by 50-60% compared to WAV with zero quality loss. - ALAC provides similar compression to FLAC but is Apple's proprietary format. - Use FLAC for archival and non-Apple playback. ## WAV: Uncompressed Standard WAV stores raw PCM audio data with minimal compression. File sizes are large (about 10 MB per minute for CD quality) but decoding is instant because there's nothing to decompress. WAV is universally supported. ## FLAC: Free Lossless Audio Codec FLAC compresses audio by 50-60% compared to WAV with zero quality loss. The original waveform is perfectly reconstructed on playback. FLAC supports metadata, album art, and up to 8 channels. ## ALAC: Apple Lossless ALAC provides similar compression to FLAC but is Apple's proprietary format. It's natively supported in iTunes, Apple Music, and iOS devices. ALAC is now open-source but sees less adoption outside the Apple ecosystem. ## Comparison | Feature | WAV | FLAC | ALAC | |---------|-----|------|------| | Compression | None | ~50-60% | ~50-60% | | Quality | Lossless | Lossless | Lossless | | Metadata | Basic | Excellent | Good | | Apple support | Yes | No (natively) | Yes | | Android support | Yes | Yes | Limited | | Streaming | No | Yes | Yes | ## Recommendation Use FLAC for archival and non-Apple playback. Use ALAC for Apple-centric workflows. Use WAV for audio editing (universal DAW compatibility) and as a master format. ### Troubleshooting Audio Sync Issues in Video Audio-video synchronization problems make content unwatchable. Learn how to diagnose and fix audio drift, delay, and sync offset issues. - URL: https://peasyaudio.com/guides/troubleshooting-audio-sync-video/ - Category: Troubleshooting - Reading time: 1 min - Words: 197 Key takeaways: - Watch for visual cues: lip movements that don't match speech, sound effects that lag behind their visual triggers, or echo-like effects where audio from a previous scene overlaps the current one. - Audio is consistently early or late by the same amount throughout. - Audio starts in sync but gradually falls out of sync, getting worse over time. - Variable frame rate (VFR) video from phones and screen recorders. ## Detecting Sync Issues Watch for visual cues: lip movements that don't match speech, sound effects that lag behind their visual triggers, or echo-like effects where audio from a previous scene overlaps the current one. ## Constant Offset ### Symptoms Audio is consistently early or late by the same amount throughout. ### Cause Different recording start times for audio and video, or a fixed delay introduced during capture or encoding. ### Solution Shift the entire audio track forward or backward by the offset amount. Use a clap or visual cue to measure the exact offset. ## Progressive Drift ### Symptoms Audio starts in sync but gradually falls out of sync, getting worse over time. ### Cause Different sample rates between audio and video (e.g., audio at 48000 Hz paired with video at 48048 Hz), or dropped frames. ### Solution Resample the audio to match the video's expected sample rate. For dropped frames, re-encode the video at a constant frame rate. ## After-Export Sync Loss ### Cause Variable frame rate (VFR) video from phones and screen recorders. Many editors struggle with VFR, creating sync issues on export. ### Solution Convert VFR to constant frame rate (CFR) before editing. ### Audio for Podcasting: Recording and Production Best Practices Professional podcast audio doesn't require expensive equipment. Learn recording techniques, editing workflow, and export settings for podcasts. - URL: https://peasyaudio.com/guides/audio-podcasting-best-practices/ - Category: Best Practice - Reading time: 1 min - Words: 173 Key takeaways: - The room matters more than the microphone. - Keep a consistent 6-8 inch distance from the microphone. - Record at 44.1 kHz, 24-bit for the best quality-to-size ratio. - Export as MP3 at 128 kbps for mono voice podcasts or 192 kbps for stereo content with music. - Never record at 16-bit — the extra headroom from 24-bit prevents clipping on loud passages. ## Room Setup The room matters more than the microphone. Record in a small, carpeted room with soft furnishings that absorb reflections. Avoid large empty rooms, kitchens, and bathrooms where sound bounces off hard surfaces. ## Microphone Technique Keep a consistent 6-8 inch distance from the microphone. Speak across the mic (slightly off-axis) rather than directly into it to reduce plosives. Use a pop filter for additional plosive protection. ## Recording Settings Record at 44.1 kHz, 24-bit for the best quality-to-size ratio. 48 kHz is also acceptable. Never record at 16-bit — the extra headroom from 24-bit prevents clipping on loud passages. ## Editing Workflow 1. Remove obvious mistakes and long pauses. 2. Reduce background noise (gentle, not aggressive). 3. Apply compression to even out volume levels. 4. Normalize to -16 LUFS (podcast standard). 5. Add intro/outro music and transitions. ## Export Settings Export as MP3 at 128 kbps for mono voice podcasts or 192 kbps for stereo content with music. Include ID3 tags (title, artist, artwork) for proper display in podcast apps. ### How to Record High-Quality Audio in Noisy Environments Capture clean audio recordings in challenging acoustic environments using noise reduction techniques and mic placement. - URL: https://peasyaudio.com/guides/how-to-record-audio-noisy-environments/ - Category: How-To - Reading time: 2 min - Words: 305 ## Recording in Noisy Environments Professional-quality audio recording often happens outside controlled studio environments — in offices, cafes, outdoor locations, and home studios with ambient noise. The right techniques and settings minimize noise capture at the source, which always produces better results than post-processing cleanup. ### Microphone Selection and Placement Dynamic microphones reject more ambient noise than condenser microphones because they're less sensitive. Cardioid polar patterns pick up sound from the front while rejecting rear noise. Get the microphone as close to the sound source as possible — halving the distance quadruples the signal-to-noise ratio. Use a pop filter for close-miking vocals. ### Room Treatment vs Post-Processing Simple room treatment beats sophisticated post-processing. Hang thick blankets or towels behind the speaker and on reflective surfaces. Close windows and doors. Turn off fans, air conditioning, and appliances with motors. Record a 10-second "room tone" (silence in the recording environment) for use as a noise profile in post-processing. ### Recording Settings Record at 24-bit depth to maximize dynamic range — this gives you 48dB more headroom than 16-bit. Set levels so peaks hit around -12dB, leaving margin for loud moments. 44.1kHz sample rate is sufficient for speech; use 48kHz for music or content that will be mixed with video (video standard is 48kHz). ### Post-Processing Noise Reduction Modern noise reduction algorithms analyze a noise sample (the room tone you recorded) and subtract its spectral profile from the entire recording. Apply noise reduction before any other processing (compression, EQ). Use moderate settings — aggressive noise reduction creates audible artifacts (underwater sound, metallic ringing). ### When to Re-Record If the signal-to-noise ratio is below 20dB (noise is clearly audible during speech), no amount of post-processing will produce acceptable results. Re-record in a quieter environment. It's faster to record well than to spend hours trying … ### Audio Bitrate and Quality Settings Guide Choose the right audio bitrate and encoding settings for music, podcasts, voice, and streaming applications. - URL: https://peasyaudio.com/guides/audio-bitrate-quality-settings-guide/ - Category: Comparison - Reading time: 2 min - Words: 325 ## Audio Bitrate and Quality Bitrate determines how much data per second is used to represent audio. Higher bitrates mean better quality but larger files. The right bitrate depends on the content type and delivery medium. ### Bitrate Recommendations by Use Case Voice/podcasts: 64-96 kbps in AAC or Opus is sufficient — speech has a limited frequency range. Music streaming: 128 kbps AAC is "good enough" for casual listening, 256 kbps AAC or 320 kbps MP3 is transparent (indistinguishable from lossless) for most listeners. Archival: lossless formats (FLAC, ALAC) preserve all audio data regardless of bitrate. ### Variable vs Constant Bitrate Constant bitrate (CBR) allocates the same data rate throughout the file. Simple passages waste bits while complex passages may sound compressed. Variable bitrate (VBR) allocates more data to complex passages and less to simple ones, producing better quality at the same average file size. VBR is preferred for most applications; CBR is needed for some streaming protocols. ### The Transparency Threshold Transparency is the point where higher bitrates produce no audible improvement. For AAC, this is typically 128-192 kbps depending on content complexity. For MP3, it's 192-256 kbps. For Opus, it's as low as 96-128 kbps for music. Testing transparency requires ABX testing (blind comparison) — most people who claim to hear differences above transparency fail controlled tests. ### Sample Rate Considerations 44.1 kHz (CD quality) captures frequencies up to 22.05 kHz — beyond the human hearing range of ~20 kHz. Higher sample rates (48, 96, 192 kHz) are useful for production (preventing aliasing during processing) but provide no audible benefit for final delivery. For podcasts and voice, 44.1 kHz is overkill — 22.05 kHz captures all speech frequencies. ### Format Selection AAC offers the best quality-per-bit for lossy compression. Opus is even better, especially at low bitrates, but … ### Audio Mastering for Podcast Distribution Podcast audio needs to meet platform loudness standards and sound consistent across episodes. Learn the mastering chain for professional podcast quality. - URL: https://peasyaudio.com/guides/audio-mastering-podcast-distribution/ - Category: Best Practice - Reading time: 1 min - Words: 271 ## Why Podcast Mastering Matters Listeners switch between podcasts on the same playlist. If your episode is significantly louder or quieter than others, listeners will either reach for the volume knob or skip your show. Mastering ensures consistent loudness and clarity across all episodes and platforms. ### The Mastering Chain Apply these processing steps in order: 1) EQ to correct any tonal imbalances from recording (reduce boominess around 200-400 Hz, add clarity at 3-5 kHz). 2) Compression to reduce dynamic range — a 3:1 ratio with medium attack and release works for speech. 3) Limiting to catch any remaining peaks and bring the overall loudness up. 4) Loudness normalization to the target standard. ### Platform Loudness Standards Apple Podcasts targets -16 LUFS (Loudness Units Full Scale). Spotify normalizes to -14 LUFS. YouTube targets -14 LUFS. The safest target is -16 LUFS — platforms that normalize to a louder target will increase your volume slightly, which is better than being turned down (which can introduce artifacts). ### Mono vs Stereo Most podcast content should be delivered in mono. Two reasons: it halves the file size, and stereo information in voice recordings adds no value for listeners using a single earbud or car speaker. Exception: podcasts with music segments or spatial audio design elements benefit from stereo. ### File Format and Quality MP3 at 128 kbps mono (or 192 kbps stereo) is the standard for podcast distribution. AAC at 96-128 kbps provides better quality at the same file size but has slightly less universal support. Include ID3 tags: title, artist, album (podcast name), track number (episode number), genre (Podcast), and artwork (3000x3000 JPEG). ### How to Concatenate and Merge Audio Files Join multiple audio files into one with proper crossfading, level matching, and format handling. - URL: https://peasyaudio.com/guides/how-to-concatenate-merge-audio-files/ - Category: How-To - Reading time: 1 min - Words: 284 ## Merging Audio Files Combining audio files is common for podcast production (intro + segments + outro), music compilation (album assembly), and audiobook creation (chapter merging). The quality of the merge depends on level matching, crossfading, and format consistency. ### Format Consistency Convert all input files to the same format, sample rate, and channel count before merging. Mixing 44.1kHz and 48kHz files produces pitch artifacts. Mixing stereo and mono files produces unexpected panning. Convert everything to the highest quality source format, then export the merged file in your target format. ### Level Matching Different recordings have different loudness levels. A quiet interview followed by a loud music bumper creates a jarring listening experience. Apply loudness normalization to each segment before merging, targeting a consistent LUFS level: -16 LUFS for podcasts (per Apple and Spotify recommendations), -14 LUFS for music. ### Crossfading Joining clips with hard cuts produces audible pops and unnatural transitions. A short crossfade (50-200ms) smooths transitions for continuous content. Longer crossfades (1-3 seconds) work for music medleys and segment transitions. Use equal-power crossfades rather than linear fades for consistent perceived volume during the transition. ### Browser-Based Audio Merging Client-side audio tools can concatenate files without server uploads. Upload your audio segments in order, specify crossfade duration, normalize levels, and export the merged result. Processing happens in your browser using the Web Audio API. For large files (podcasts, audiobooks), this may take a few minutes depending on your device. ### Metadata for Merged Files Update the metadata of the merged file: total duration, chapter markers (if supported by the format), title, and artist. For audiobooks, M4B format supports chapter markers. For podcasts, add ID3 tags with episode title, podcast name, and episode number. ### Troubleshooting Video Export and Rendering Errors Fix common video export failures including codec errors, audio sync issues, and rendering crashes. - URL: https://peasyaudio.com/guides/troubleshooting-video-export-rendering-errors/ - Category: Troubleshooting - Reading time: 2 min - Words: 315 ## Troubleshooting Video Export Video export failures are frustrating because they often occur after a long rendering process. Understanding common failure modes helps you diagnose and fix issues without starting from scratch. ### Out of Memory Errors Video rendering is memory-intensive, especially at high resolutions with effects. 4K video with complex effects can consume 8GB+ of RAM. Solutions: close other applications during rendering, reduce preview quality (render quality can stay high), render in sections and concatenate afterward, or use proxy editing (edit with low-res proxies, render from full-res originals). ### Audio/Video Sync Issues Sync problems usually originate from variable frame rate (VFR) source footage. Most screen recorders and smartphones record in VFR, where the frame rate varies based on scene complexity. Convert VFR to constant frame rate (CFR) before editing. If sync drift is gradual (audio slowly gets ahead or behind), the frame rate of the timeline doesn't match the source. ### Codec Compatibility The export codec must be supported by your target platform. H.264 is the safest choice for universal compatibility. H.265 (HEVC) produces smaller files but isn't supported everywhere. ProRes is preferred for professional editing workflows but produces very large files. AV1 offers the best compression but encoding is extremely slow. ### Color Shift After Export If exported video looks different from the preview — washed out, oversaturated, or with a color cast — check the color space settings. Editing in Rec. 709 and exporting in Rec. 2020 (or vice versa) causes visible color shifts. Ensure the export color space matches the source and editing color space. ### Partial Render Failures If rendering fails at a specific timestamp, that section likely contains a problematic element: a corrupt source clip, an unsupported effect, or a transition that exceeds available memory. Navigate to the failure point, simplify or remove the … ### How to Remove Echo and Reverb from Audio Recordings Recording in reflective rooms produces distracting echo and reverb. Learn the techniques and tools for reducing room acoustics in post-production. - URL: https://peasyaudio.com/guides/how-to-remove-echo-reverb-audio/ - Category: How-To - Reading time: 1 min - Words: 222 ## Understanding the Problem Echo is a distinct repetition of sound reflected off hard surfaces. Reverb is many reflections blending together, creating a "roomy" or "bathroom" quality. Both degrade speech clarity and make audio sound unprofessional. ### Prevention is Better Than Cure No amount of post-processing fully removes reverb without artifacts. Treat your recording space first: hang blankets or acoustic panels on hard walls, place a thick rug on hard floors, close windows and doors, and record away from walls. A closet full of clothes is an excellent improvised recording booth. ### De-Reverb Processing Modern de-reverb algorithms analyze the reverb characteristics (decay time, frequency profile) and subtract them from the signal. The process works best on voice recordings with clear speech. Set the reduction amount conservatively — aggressive de-reverb introduces "metallic" or "underwater" artifacts that sound worse than mild reverb. ### Frequency-Specific Treatment Reverb is most problematic in the 200-800 Hz range where it muddies speech. Apply de-reverb processing more aggressively in this range while treating higher frequencies more gently. Follow up with a gentle high-pass filter at 80-100 Hz to remove low-frequency rumble that often accompanies reverb. ### Workflow Order Apply de-reverb processing before other effects. Processing order matters: de-reverb first, then noise reduction, then EQ, then compression. Applying compression before de-reverb amplifies the reverb tail, making it harder to remove. ### Digital Audio Workstation vs Online Audio Tools Compare full DAWs with browser-based audio tools for different editing needs from simple trims to complex production. - URL: https://peasyaudio.com/guides/daw-vs-online-audio-tools-comparison/ - Category: Comparison - Reading time: 2 min - Words: 316 ## DAW vs Online Audio Tools Desktop Digital Audio Workstations (DAWs) and browser-based audio tools serve different segments of audio editing needs. Understanding where each excels helps you choose the right tool without over-investing in complexity you don't need. ### When Online Tools Are Enough Browser-based tools handle: trimming and cutting, format conversion, basic noise reduction, loudness normalization, merging files, extracting audio from video, and simple effects (fade in/out, speed change). If your editing is mostly cut-trim-export, online tools save the installation, learning curve, and cost of a DAW. Processing happens locally in your browser, maintaining privacy. ### When You Need a DAW Multi-track mixing (voice + music + sound effects), advanced noise reduction with real-time preview, time-stretching without pitch change, VST plugin support for professional effects, MIDI editing for music production, and projects that require saving and resuming work across sessions. DAWs are necessary when audio production is a core part of your workflow. ### Popular DAW Options GarageBand (free, macOS) — excellent for beginners, surprisingly capable. Audacity (free, cross-platform) — powerful but dated interface. Reaper ($60) — professional quality at a fraction of pro DAW costs. Logic Pro ($200, macOS) — industry standard for music production. Pro Tools (subscription) — film and broadcast industry standard. ### Hybrid Workflows Many professionals use online tools for quick tasks and DAWs for complex projects. Convert a batch of files to a consistent format using a browser tool, then import into the DAW for multi-track editing. Export from the DAW, then use an online tool for final format conversion and loudness normalization for different distribution platforms. ### Performance Considerations DAWs consume significant CPU during real-time effects processing and can struggle with many simultaneous tracks on older hardware. Browser-based tools process offline (not in real-time), so they can handle operations that would strain … ### Audio File Formats for Music Production Music production requires understanding the strengths and tradeoffs of different audio formats at each stage of the workflow — recording, editing, mixing, and distribution. - URL: https://peasyaudio.com/guides/audio-formats-music-production/ - Category: Comparison - Reading time: 1 min - Words: 296 Key takeaways: - Always record in an uncompressed lossless format. - ## Recording Stage Always record in an uncompressed lossless format. - Never convert to a lossy format and back — each lossy encoding cycle permanently removes data. - Always generate lossy formats from the lossless master, never from another lossy format. ## Recording Stage Always record in an uncompressed lossless format. WAV (PCM) at 24-bit/48 kHz is the industry standard. 24-bit provides 144 dB of dynamic range, eliminating the need to record "hot" (close to 0 dB). 48 kHz is the standard for video-synced audio; 44.1 kHz is standard for music-only production. Higher sample rates (96 kHz, 192 kHz) offer marginal benefits for most recording scenarios. ### Editing and Mixing Keep everything in WAV or AIFF during editing and mixing. Never convert to a lossy format and back — each lossy encoding cycle permanently removes data. Your DAW's project file references these lossless files; the project itself stores mix settings, plugin parameters, and automation data. ### Mastering and Delivery The master file should be WAV 24-bit at the project sample rate. From this master, create distribution formats: WAV 16-bit/44.1 kHz for CD, FLAC for lossless digital distribution, MP3 320 kbps for compatibility, and AAC 256 kbps for iTunes/Apple Music. Always generate lossy formats from the lossless master, never from another lossy format. ### Format Comparison for Distribution | Format | Quality | File Size | Compatibility | |--------|---------|-----------|---------------| | WAV 16/44.1 | Lossless | ~10 MB/min | Universal | | FLAC | Lossless | ~6 MB/min | Excellent | | ALAC | Lossless | ~6 MB/min | Apple ecosystem | | MP3 320 | Lossy (excellent) | ~2.5 MB/min | Universal | | AAC 256 | Lossy (excellent) | ~2 MB/min | Excellent | | OGG 192 | Lossy (good) | ~1.5 MB/min | Good | ### Metadata Embed metadata in all distribution files: artist, title, album, track number, genre, year, and artwork. Use a consistent naming convention for files: `Artist - Album - TrackNumber - Title.ext`. For streaming platforms, ensure metadata matches your distributor's requirements exactly. ### Audio Equalization Fundamentals for Content Creators Master EQ techniques to improve voice clarity, remove muddiness, and achieve professional podcast audio. - URL: https://peasyaudio.com/guides/audio-equalization-fundamentals-content-creators/ - Category: How-To - Reading time: 1 min - Words: 255 Key takeaways: - Equalization (EQ) is the process of adjusting the volume of specific frequency ranges in an audio signal. - The fundamental frequency of the human voice ranges from 85-255 Hz. - Apply a high-pass filter at 80-100 Hz to remove rumble and handling noise. - Over-boosting presence frequencies (3-5 kHz) creates harsh, fatiguing audio. - No amount of EQ can fix a bad recording environment. ## Understanding Equalization Equalization (EQ) is the process of adjusting the volume of specific frequency ranges in an audio signal. For content creators, EQ is the most important tool for achieving clear, professional-sounding voice recordings. Understanding frequency ranges and their effect on voice quality is essential. ## Voice Frequency Ranges The fundamental frequency of the human voice ranges from 85-255 Hz. Clarity and intelligibility come from the 1-4 kHz range. Presence and air live above 5 kHz. The 200-500 Hz range often contains muddiness, while 2-4 kHz adds harshness if boosted too aggressively. ## Essential EQ Moves for Voice Apply a high-pass filter at 80-100 Hz to remove rumble and handling noise. Cut 2-3 dB around 200-300 Hz to reduce muddiness (the proximity effect from close-miking). Boost 1-2 dB around 2-3 kHz for clarity. Add a gentle shelf boost above 8 kHz for air and presence. Always cut before you boost — subtractive EQ sounds more natural. ## Common Mistakes Over-boosting presence frequencies (3-5 kHz) creates harsh, fatiguing audio. Scooping too much low-mid content removes warmth and body. Using narrow Q values on voice creates unnatural resonances. Not comparing with EQ bypassed leads to the louder-is-better illusion — match perceived loudness when A/B testing. ## Room Acoustics and EQ No amount of EQ can fix a bad recording environment. Room reflections create comb filtering that EQ cannot address. Invest in acoustic treatment (absorption panels, bass traps) before reaching for EQ. Record test samples in your space and analyze the frequency response to identify room-induced problems. ### Audio Compression vs Limiting: When to Use Each Compare dynamic range compression and limiting for different audio production scenarios. - URL: https://peasyaudio.com/guides/audio-compression-vs-limiting-comparison/ - Category: Comparison - Reading time: 1 min - Words: 268 Key takeaways: - Both compression and limiting reduce dynamic range — the difference between the loudest and quietest parts of audio. - Compression is ideal for evening out vocal performances where volume varies naturally. - Limiting serves as a safety net — preventing audio from exceeding a maximum level. - Over-compressing removes all dynamic expression, creating lifeless, fatiguing audio. - For podcast vocals, start with a compressor at 3:1 ratio, -18 dB threshold, 10ms attack, 100ms release. ## Compression and Limiting Defined Both compression and limiting reduce dynamic range — the difference between the loudest and quietest parts of audio. A compressor applies gradual gain reduction above a threshold with adjustable ratio (typically 2:1 to 8:1). A limiter is essentially a compressor with an extreme ratio (10:1 or higher), creating a hard ceiling. ## When to Use Compression Compression is ideal for evening out vocal performances where volume varies naturally. A podcast host who speaks softly then laughs loudly benefits from 3:1 compression. Music mixing uses compression to add sustain to instruments, glue mix elements together, and control transients. Moderate compression (2-4 dB gain reduction) is usually transparent. ## When to Use Limiting Limiting serves as a safety net — preventing audio from exceeding a maximum level. During mastering, a limiter raises overall loudness while preventing clipping. For broadcast and streaming, limiters ensure compliance with loudness standards (LUFS targets). Live sound uses limiters to protect speakers from damage. ## Common Mistakes Over-compressing removes all dynamic expression, creating lifeless, fatiguing audio. Setting attack times too fast on a limiter destroys transients, making audio sound pumpy. Stacking multiple compressors without understanding gain staging leads to noise amplification. Not using makeup gain after compression results in quieter overall levels. ## Practical Setup Guide For podcast vocals, start with a compressor at 3:1 ratio, -18 dB threshold, 10ms attack, 100ms release. Follow with a limiter set at -1 dB ceiling. Adjust the compressor threshold until you see 3-6 dB of gain reduction on average. Monitor your output level — target -16 LUFS for podcasts, -14 LUFS for music streaming. ### Video Accessibility: Audio Descriptions and Transcripts Making video content accessible requires more than captions. Learn about audio descriptions for blind viewers, transcripts, and accessible player controls. - URL: https://peasyaudio.com/guides/video-accessibility-audio-descriptions/ - Category: Best Practice - Reading time: 1 min - Words: 233 Key takeaways: - Captions serve deaf and hard-of-hearing viewers, but blind and low-vision viewers also need accommodations. - ### Audio Descriptions Audio descriptions are narrations inserted during natural pauses in dialogue, describing important visual elements: on-screen text, facial expressions, scene changes, and actions. - ### Accessible Video Players The video player itself must be keyboard-navigable with visible focus indicators. - Screen readers must be able to announce the current playback state and available controls. ## Beyond Captions Captions serve deaf and hard-of-hearing viewers, but blind and low-vision viewers also need accommodations. Audio descriptions narrate visual information that isn't conveyed through dialogue. Transcripts provide a text alternative that works for both groups and is searchable by machines. ### Audio Descriptions Audio descriptions are narrations inserted during natural pauses in dialogue, describing important visual elements: on-screen text, facial expressions, scene changes, and actions. For example: "Sarah glances at the document on her desk, then looks out the window at the approaching storm." Extended audio descriptions pause the video to allow longer narrations when natural pauses are insufficient. ### Transcripts A complete transcript includes dialogue, speaker identification, sound effects, music descriptions, and descriptions of significant visual content. Transcripts serve multiple purposes: accessibility, SEO (search engines can index the text), content repurposing (blog posts, social media), and translation. ### Accessible Video Players The video player itself must be keyboard-navigable with visible focus indicators. Play/pause, volume, progress bar, caption toggle, and fullscreen controls must all be operable without a mouse. Screen readers must be able to announce the current playback state and available controls. ### WCAG Requirements Level A requires captions for pre-recorded video and a text alternative (transcript or audio description). Level AA requires captions for live video and audio descriptions for pre-recorded video. Level AAA requires sign language interpretation and extended audio descriptions. At minimum, aim for Level AA compliance. ### Removing Background Noise from Audio Recordings Techniques for cleaning up audio by removing hum, hiss, and environmental noise effectively. - URL: https://peasyaudio.com/guides/removing-background-noise-audio-recordings/ - Category: Troubleshooting - Reading time: 1 min - Words: 271 Key takeaways: - Background noise falls into two categories: stationary noise (consistent hum, hiss, fan noise) and non-stationary noise (traffic, voices, clicks). - A noise gate silences audio below a threshold, effectively muting quiet sections where noise is most audible. - For removing noise that's present even during speech, spectral noise reduction analyzes a noise profile (a sample of pure noise) and subtracts those frequencies from the entire recording. - For 50/60 Hz hum (electrical interference), use a notch filter at the fundamental frequency and its harmonics (120 Hz, 180 Hz, etc.). - The best noise removal happens before recording. ## Types of Background Noise Background noise falls into two categories: stationary noise (consistent hum, hiss, fan noise) and non-stationary noise (traffic, voices, clicks). Stationary noise is easier to remove because its frequency profile is predictable. Non-stationary noise requires more sophisticated techniques. ## Noise Gate Basics A noise gate silences audio below a threshold, effectively muting quiet sections where noise is most audible. Set the threshold just above the noise floor. Use a fast attack (1-5ms) so speech isn't clipped, and a medium release (50-200ms) to avoid choppy cutoffs. Noise gates don't remove noise during speech — they only silence pauses. ## Spectral Noise Reduction For removing noise that's present even during speech, spectral noise reduction analyzes a noise profile (a sample of pure noise) and subtracts those frequencies from the entire recording. Capture at least 2 seconds of room noise before recording. Apply reduction conservatively — aggressive settings create underwater-sounding artifacts. ## Dealing with Specific Noise Types For 50/60 Hz hum (electrical interference), use a notch filter at the fundamental frequency and its harmonics (120 Hz, 180 Hz, etc.). For air conditioning rumble, apply a high-pass filter at 100-150 Hz. For computer fan noise, spectral reduction works well since fans produce consistent broadband noise. For mouth clicks, use a de-clicker plugin or manually edit them out. ## Prevention Over Cure The best noise removal happens before recording. Use a dynamic microphone in noisy environments (they reject more room noise than condensers). Position the mic close to the source (6-12 inches). Turn off unnecessary electronics. Record during quiet hours. A $20 moving blanket draped over a frame reduces room reflections dramatically. ### Audio File Formats: Lossy vs Lossless Compared Compare MP3, AAC, FLAC, WAV, and OGG for quality, file size, and compatibility. - URL: https://peasyaudio.com/guides/audio-file-formats-lossy-lossless-compared/ - Category: Comparison - Reading time: 1 min - Words: 257 Key takeaways: - Lossy formats (MP3, AAC, OGG Vorbis) permanently discard audio data to achieve small file sizes. - MP3 (MPEG-1 Audio Layer III) is supported by virtually every device and platform. - AAC (Advanced Audio Coding) generally produces better quality than MP3 at the same bitrate. - FLAC typically achieves 50-60% compression compared to WAV while preserving bit-perfect audio. - For music distribution and streaming, use AAC or MP3 at 256-320 kbps. ## The Lossy vs Lossless Divide Lossy formats (MP3, AAC, OGG Vorbis) permanently discard audio data to achieve small file sizes. Lossless formats (FLAC, ALAC, WAV) preserve every sample of the original recording. The choice depends on your priorities: storage and bandwidth versus audio fidelity. ## MP3: Universal Compatibility MP3 (MPEG-1 Audio Layer III) is supported by virtually every device and platform. At 320 kbps, MP3 is transparent (indistinguishable from lossless) for most listeners. At 128 kbps, artifacts become audible on good speakers. MP3's main weakness is its age — newer codecs achieve better quality at the same bitrate. ## AAC: The Modern Standard AAC (Advanced Audio Coding) generally produces better quality than MP3 at the same bitrate. It's the default format for Apple devices, YouTube, and most streaming services. AAC at 256 kbps is roughly equivalent to MP3 at 320 kbps. HE-AAC extends this efficiency for low-bitrate streaming applications. ## FLAC: Lossless Compression FLAC typically achieves 50-60% compression compared to WAV while preserving bit-perfect audio. It's the standard for archival and audiophile use. FLAC supports metadata and album art. The only downside is limited native support on Apple devices (though third-party players handle it well). ## Choosing the Right Format For music distribution and streaming, use AAC or MP3 at 256-320 kbps. For archival and mastering, use FLAC or WAV. For podcasts, MP3 at 128 kbps mono or 192 kbps stereo is the industry standard. For web audio, consider Opus — it outperforms all other codecs at every bitrate and is supported in all modern browsers. ### Audio Loudness Standards for Streaming Platforms Meet loudness requirements for Spotify, YouTube, Apple Music, and podcast platforms. - URL: https://peasyaudio.com/guides/audio-loudness-standards-streaming-platforms/ - Category: Best Practice - Reading time: 1 min - Words: 279 Key takeaways: - Streaming platforms normalize audio to consistent loudness levels. - Spotify normalizes to -14 LUFS (integrated), YouTube to -14 LUFS, Apple Music to -16 LUFS, and Tidal to -14 LUFS. - LUFS (Loudness Units Full Scale) measures perceived loudness over time. - First, mix your audio without worrying about loudness. - The loudness war mentality of making everything as loud as possible is counterproductive on streaming platforms — louder masters just get turned down more. ## Why Loudness Standards Matter Streaming platforms normalize audio to consistent loudness levels. If your audio is louder than the target, it gets turned down — and any distortion from over-limiting remains. If it's too quiet, it gets turned up — but noise and dynamic range issues become more apparent. Mastering to the correct target ensures your audio sounds as intended. ## Platform Targets Spotify normalizes to -14 LUFS (integrated), YouTube to -14 LUFS, Apple Music to -16 LUFS, and Tidal to -14 LUFS. Podcasts on Apple Podcasts and Spotify should target -16 LUFS with a -1 dB true peak ceiling. Broadcast standards (EBU R128) target -23 LUFS. ## Measuring Loudness LUFS (Loudness Units Full Scale) measures perceived loudness over time. Integrated LUFS averages the entire file. Short-term LUFS measures 3-second windows. Momentary LUFS uses 400ms windows. Use a loudness meter plugin during mastering — most DAWs include one, or use the free Youlean Loudness Meter. ## Mastering Workflow First, mix your audio without worrying about loudness. Then, apply a limiter with the ceiling at -1 dB true peak. Gradually increase the limiter's input gain until the integrated LUFS reads your target. If you need more than 3-4 dB of limiting to reach the target, go back and adjust your mix levels rather than pushing the limiter harder. ## Common Pitfalls The loudness war mentality of making everything as loud as possible is counterproductive on streaming platforms — louder masters just get turned down more. Mono compatibility issues become apparent at lower playback levels. Always check your master in mono and at low volumes. True peak limiting is essential — inter-sample peaks can cause clipping in lossy codec conversion. ### How to Batch Process Multiple File Conversions Converting files one at a time is tedious and error-prone. Learn strategies for batch converting documents, images, audio, and video efficiently. - URL: https://peasyaudio.com/guides/how-to-batch-process-file-conversions/ - Category: How-To - Reading time: 1 min - Words: 286 ## The Batch Conversion Workflow Whether you're converting 50 images from HEIC to JPEG, 200 documents from DOCX to PDF, or an audio library from FLAC to MP3, the workflow is the same: organize inputs, configure conversion settings, process in batch, verify outputs, and handle failures. ### Organizing Input Files Start by collecting all source files in a single directory structure. Verify file types — not all files with the same extension are the same format (a .doc file might be Word 97 or Word 2003 format). Count files and note total size to estimate processing time and storage requirements. ### Consistent Settings Define your conversion settings once and apply them uniformly. For images: target format, quality level, maximum dimensions, color profile. For documents: output format, font embedding, image compression. For audio: codec, bitrate, sample rate, mono/stereo. Document these settings for reproducibility. ### Error Recovery Large batches will encounter failures — corrupted source files, unsupported features, and edge cases. Design your workflow to: log each file's processing result (success/failure/warning), continue processing after failures (don't stop on first error), and generate a summary report. Keep failed files in a separate output directory for manual review. ### Output Verification Spot-check converted files at regular intervals (every 10th file, for example). Compare file sizes — dramatically smaller or larger outputs may indicate conversion errors. For images, do visual spot checks. For documents, verify formatting preservation. For audio/video, check duration matches and listen/watch brief samples. ### Naming and Organization Maintain source file names in the output, changing only the extension. Create a parallel directory structure if the source is organized in folders. Include a conversion log file with each output batch documenting source files, settings used, and any issues encountered. ### Comparing Audio Compression: Lossy vs Lossless Formats Choosing between lossy and lossless audio formats involves trade-offs between file size, sound quality, and compatibility. This comparison breaks down the technical differences and helps you pick the right format for your needs. - URL: https://peasyaudio.com/guides/comparing-audio-compression-lossy-vs-lossless/ - Category: Comparison - Reading time: 3 min - Words: 524 Key takeaways: - Digital audio files store thousands of samples per second — typically 44,100 (CD quality) or 48,000 (professional standard) samples per second, with each sample using 16 or 24 bits. - Lossy codecs use psychoacoustic models to identify and discard audio data that most humans can't perceive. - Lossless codecs compress audio data without discarding any information. - WAV: ~10MB, FLAC: ~5-7MB, MP3 320kbps: ~2.4MB, MP3 128kbps: ~960KB, AAC 256kbps: ~1.9MB, Opus 160kbps: ~1.2MB. - For web-based audio tools, format support varies. ## How Audio Compression Works Digital audio files store thousands of samples per second — typically 44,100 (CD quality) or 48,000 (professional standard) samples per second, with each sample using 16 or 24 bits. A single minute of uncompressed stereo CD-quality audio takes about 10MB. Compression reduces this size through two fundamentally different approaches. ## Lossy Compression ### How It Works Lossy codecs use psychoacoustic models to identify and discard audio data that most humans can't perceive. This includes frequencies masked by louder nearby frequencies, sounds below the threshold of hearing, and temporal masking where a loud sound makes the ear temporarily less sensitive. ### Common Lossy Formats **MP3** remains the most universally compatible lossy format. At 320kbps, MP3 is transparent (indistinguishable from the original) for most listeners. At 128kbps, artifacts become noticeable on good headphones. **AAC** (Advanced Audio Coding) generally provides better quality than MP3 at the same bitrate. It's the default format for Apple devices, YouTube, and many streaming services. At 256kbps, AAC matches MP3 320kbps quality. **OGG Vorbis** is an open-source alternative that offers excellent quality-to-size ratio. It's well supported in web browsers and games but less common on portable devices. **Opus** is the newest and most efficient lossy codec. It excels at both speech and music, and provides transparent quality at bitrates as low as 160kbps. It's the standard for WebRTC and Discord. ## Lossless Compression ### How It Works Lossless codecs compress audio data without discarding any information. The decompressed output is bit-for-bit identical to the original. They achieve this through predictive modeling (predicting each sample based on previous samples) and entropy coding (representing common patterns with shorter codes). ### Common Lossless Formats **FLAC** (Free Lossless Audio Codec) is the dominant open-source lossless format. It typically compresses to 50-70% of the original WAV file size. … ### Audio Noise Reduction: Techniques for Clean Recordings Background noise in audio recordings — hum, hiss, traffic, and room ambience — degrades listener experience and professionalism. Modern noise reduction techniques can clean recordings without introducing distracting artifacts. - URL: https://peasyaudio.com/guides/audio-noise-reduction-techniques/ - Category: How-To - Reading time: 1 min - Words: 293 Key takeaways: - Consistent, predictable noise that does not change over time: air conditioning hum (60/120 Hz in the US, 50/100 Hz in Europe), computer fan noise, tape hiss, electrical buzz. - The classic noise reduction technique. - A noise gate mutes audio below a threshold level. - Electrical hum at 50/60 Hz (and harmonics at 100/120, 150/180 Hz) is removed with notch filters tuned to the exact frequencies. ## Types of Audio Noise ### Stationary Noise Consistent, predictable noise that does not change over time: air conditioning hum (60/120 Hz in the US, 50/100 Hz in Europe), computer fan noise, tape hiss, electrical buzz. Easiest to remove because algorithms can learn the pattern. ### Non-Stationary Noise Intermittent, unpredictable noise: traffic, construction, dog barking, keyboard clicks, phone notifications. Harder to remove because the pattern changes. May require manual editing (cutting affected segments) rather than algorithmic removal. ## Spectral Subtraction The classic noise reduction technique. The process: 1. Capture a 'noise profile' from a segment of pure noise (no speech or music) 2. The algorithm learns the frequency spectrum of the noise 3. It subtracts that spectrum from the entire recording Effective for stationary noise. Over-application creates a 'underwater' or 'musical noise' artifact — metallic warbling in quiet passages. ## Gate/Expander A noise gate mutes audio below a threshold level. When no one is speaking, the gate closes and the noise disappears. When speech exceeds the threshold, the gate opens. **Settings:** Set the threshold just above the noise floor. Use a fast attack (1-2 ms) so the gate opens before clipping the beginning of words. Use a moderate release (50-100 ms) so the gate does not chop off word endings. ## De-Hum Electrical hum at 50/60 Hz (and harmonics at 100/120, 150/180 Hz) is removed with notch filters tuned to the exact frequencies. Narrow Q values (high selectivity) remove the hum without affecting nearby musical content. ## Best Practice Order 1. **De-hum first** — Remove electrical noise before other processing 2. **Noise reduction** — Spectral subtraction on stationary noise 3. **Gate** — Clean up remaining low-level noise between speech segments 4. **EQ and compression** — Apply after noise reduction for the cleanest signal ### Audio Normalization and Loudness Standards (LUFS Explained) Different audio files at different volumes force listeners to constantly adjust their volume control. Loudness normalization to industry standards (LUFS) ensures consistent playback across tracks, episodes, and platforms. - URL: https://peasyaudio.com/guides/audio-normalization-lufs-guide/ - Category: Best Practice - Reading time: 2 min - Words: 359 Key takeaways: - Peak normalization adjusts audio so the loudest sample hits 0 dBFS. - LUFS (also called LKFS) measures perceived loudness using a psychoacoustic model that weights frequencies according to human hearing sensitivity. - Streaming platforms normalize audio automatically. - Loudness Range (LRA) measures the dynamic variation within a track. - Check true peak — after gain, peaks must not exceed -1 dBTP. ## Why Peak Normalization Is Wrong Peak normalization adjusts audio so the loudest sample hits 0 dBFS. This tells you nothing about perceived loudness. A whispered voice recording and a heavy metal track can both peak at 0 dBFS while sounding vastly different in loudness. Peak normalization is a leftover from the analog era and should not be used for delivery. ## LUFS: Loudness Units Full Scale LUFS (also called LKFS) measures perceived loudness using a psychoacoustic model that weights frequencies according to human hearing sensitivity. It accounts for the fact that humans perceive midrange frequencies as louder than bass or treble at the same physical level. ## Platform Loudness Standards | Platform | Target LUFS | True Peak Max | |----------|-----------|---------------| | Spotify | -14 LUFS | -1 dBTP | | Apple Music | -16 LUFS | -1 dBTP | | YouTube | -14 LUFS | -1 dBTP | | Podcasts (mono) | -16 LUFS | -1 dBTP | | Podcasts (stereo) | -19 LUFS | -1 dBTP | | Broadcast TV (US) | -24 LUFS | -2 dBTP | | Broadcast TV (EU) | -23 LUFS | -1 dBTP | ## What Happens If You Ignore LUFS Streaming platforms normalize audio automatically. If your track is mastered at -8 LUFS (common in loudness war masters), Spotify turns it down by 6 dB. The dynamic range lost to aggressive limiting cannot be restored — the track sounds smaller and less dynamic next to properly mastered music. ## How to Normalize 1. **Measure** the current integrated loudness (full track/episode) 2. **Calculate** the difference from the target (e.g., current -20 LUFS, target -14 LUFS → apply +6 dB gain) 3. **Apply gain** uniformly to the entire file 4. **Check true peak** — after gain, peaks must not exceed -1 dBTP. If … ### Audio Format Conversion: Avoiding Quality Loss and Common Pitfalls Converting between audio formats seems straightforward but introduces quality loss if done incorrectly. Understanding transcoding chains, generational loss, and format capabilities prevents common conversion mistakes. - URL: https://peasyaudio.com/guides/audio-format-conversion-best-practices/ - Category: Best Practice - Reading time: 2 min - Words: 336 Key takeaways: - Converting MP3 to AAC (or any lossy-to-lossy conversion) applies a second round of psychoacoustic compression to already-compressed audio. - When changing sample rate (e.g., 48 kHz studio → 44.1 kHz CD), use a high-quality sample rate converter (SRC). - Reducing bit depth (24-bit → 16-bit) requires dithering — adding a tiny amount of shaped noise that masks the quantization distortion from truncating sample values. - When converting formats, metadata (title, artist, album, artwork) may not transfer automatically. - When converting a large library, process a small test batch first (10 files). ## The Cardinal Rule: Never Transcode Lossy to Lossy Converting MP3 to AAC (or any lossy-to-lossy conversion) applies a second round of psychoacoustic compression to already-compressed audio. Each generation degrades quality further. The artifacts from the first compression are treated as audio data by the second encoder, producing compounding distortion. Always convert from a lossless source. ## Conversion Matrix | From → To | Quality Impact | Recommended? | |-----------|---------------|-------------| | WAV → FLAC | None (lossless) | Yes | | FLAC → WAV | None (lossless) | Yes | | WAV → MP3/AAC | One-time lossy | Yes (final delivery) | | FLAC → MP3/AAC | One-time lossy | Yes (final delivery) | | MP3 → AAC | Double lossy | No — use lossless source | | AAC → MP3 | Double lossy | No — use lossless source | | MP3 → FLAC | Wastes space, no quality gain | No | | MP3 → WAV | Wastes space, no quality gain | Only for editing | ## Sample Rate Conversion When changing sample rate (e.g., 48 kHz studio → 44.1 kHz CD), use a high-quality sample rate converter (SRC). Poor-quality SRC introduces aliasing artifacts — subtle metallic distortion. Most professional audio editors use high-quality SRC by default. ## Bit Depth Conversion Reducing bit depth (24-bit → 16-bit) requires dithering — adding a tiny amount of shaped noise that masks the quantization distortion from truncating sample values. Without dithering, quiet passages develop a harsh, granular quality. ## Metadata Preservation When converting formats, metadata (title, artist, album, artwork) may not transfer automatically. ID3 tags (MP3), Vorbis comments (FLAC/OGG), and MP4 atoms (AAC/M4A) use different metadata containers. Verify that tags survive the conversion and re-apply manually if needed. ## Batch Conversion Tips When converting a large library, process a … ### Recording Studio-Quality Audio at Home: Equipment and Technique Professional-sounding audio recordings no longer require a commercial studio. With the right microphone technique, acoustic treatment, and signal chain, home recordings can match broadcast quality at a fraction of the cost. - URL: https://peasyaudio.com/guides/recording-studio-quality-audio-home/ - Category: How-To - Reading time: 1 min - Words: 296 Key takeaways: - A $100 microphone in a well-treated room sounds better than a $2,000 microphone in an untreated room. - Hard, flat surfaces (walls, desk, ceiling) reflect sound waves that arrive at the microphone milliseconds after the direct sound, creating a 'roomy' or 'echoey' quality. - Position the microphone 15-20 cm (6-8 inches) from the mouth. - Microphone → Pop filter → Shock mount → Audio interface → Computer. ## The Three Pillars of Good Audio In order of importance: 1. **Room acoustics** (most impactful, cheapest to improve) 2. **Microphone technique** (positioning and distance) 3. **Equipment** (least impactful above a minimum threshold) A $100 microphone in a well-treated room sounds better than a $2,000 microphone in an untreated room. Invest in acoustics first. ## Room Treatment Basics ### What to Treat Hard, flat surfaces (walls, desk, ceiling) reflect sound waves that arrive at the microphone milliseconds after the direct sound, creating a 'roomy' or 'echoey' quality. Absorption panels on the wall behind the speaker and at the first reflection points (the walls to the left and right of the speaker) eliminate the worst reflections. ### Budget Solutions - Heavy blankets hung on walls (surprisingly effective) - Bookshelves filled with books (natural diffusion) - Recording inside a closet full of clothes (excellent absorption) - Portable vocal shields behind the microphone ## Microphone Technique ### Distance Position the microphone 15-20 cm (6-8 inches) from the mouth. Too close causes proximity effect (boomy bass buildup) and plosive pops. Too far captures more room reflections and reduces the signal-to-noise ratio. ### Angle Speak slightly off-axis (15-30 degrees to the side) rather than directly into the microphone capsule. This reduces plosives (p, b, t sounds) and sibilance (s, sh sounds) while maintaining a full, natural tone. ### Pop Filter A pop filter (mesh screen between mouth and microphone) is non-negotiable for speech recording. It blocks the burst of air from plosive consonants that would otherwise create a loud, low-frequency 'pop' in the recording. ## Signal Chain Microphone → Pop filter → Shock mount → Audio interface → Computer. Set the interface gain so normal speaking voice peaks at -12 to -6 dBFS. This leaves headroom for loud moments without clipping. ### Browser Fingerprinting: How Websites Track You Without Cookies Discover how browser fingerprinting works, what data points websites collect, and practical countermeasures to reduce your trackability. Covers canvas fingerprinting, WebGL hashing, and audio context analysis. - URL: https://peasyaudio.com/guides/browser-fingerprinting-tracking-explained/ - Category: How-To - Reading time: 1 min - Words: 229 Key takeaways: - While cookies require user consent in many jurisdictions, browser fingerprinting operates silently. - Canvas fingerprinting renders invisible text or graphics, then reads the pixel data. - Research from the EFF's Panopticlick project showed that over 83% of browsers have a unique fingerprint. - Complete fingerprint resistance is extremely difficult, but you can reduce uniqueness significantly: ## Beyond Cookies While cookies require user consent in many jurisdictions, browser fingerprinting operates silently. Websites collect dozens of data points from your browser — screen resolution, installed fonts, WebGL renderer, timezone, language preferences, and hardware capabilities. Combined, these create a unique identifier that persists across sessions without storing anything on your device. ## Common Fingerprinting Techniques **Canvas fingerprinting** renders invisible text or graphics, then reads the pixel data. Subtle rendering differences between GPUs, font engines, and antialiasing algorithms produce a unique hash. **WebGL fingerprinting** queries the GPU model, vendor, and rendering quirks. **AudioContext fingerprinting** processes audio signals and compares the output — different hardware produces measurably different results. ## How Unique Are You? Research from the EFF's Panopticlick project showed that over 83% of browsers have a unique fingerprint. Even users who block cookies and use private browsing can be tracked. The more extensions, fonts, and customizations you have, the more distinctive your browser becomes. ## Countermeasures Complete fingerprint resistance is extremely difficult, but you can reduce uniqueness significantly: - Use Firefox with `privacy.resistFingerprinting` enabled - Tor Browser provides the strongest anti-fingerprinting protections - Avoid installing unusual browser extensions - Use standard screen resolutions and system fonts - Disable JavaScript on sites that do not require it Test your current fingerprint exposure with the Peasy browser fingerprint analyzer to see exactly which data points make your browser unique. ### Audio Metadata and ID3 Tags: A Complete Reference Audio metadata — embedded information about title, artist, album, artwork, and more — determines how music appears in players, libraries, and streaming services. Proper tagging prevents misidentified tracks and missing artwork. - URL: https://peasyaudio.com/guides/audio-metadata-id3-tags-guide/ - Category: How-To - Reading time: 2 min - Words: 342 ## Metadata Standards by Format | Format | Metadata Container | Common Fields | |--------|-------------------|---------------| | MP3 | ID3v2.3 / ID3v2.4 | Title, Artist, Album, Year, Track, Genre, Art | | FLAC | Vorbis Comments | Same fields, different container | | AAC/M4A | MP4 Atoms (iTunes-style) | Same fields, plus Apple-specific tags | | OGG | Vorbis Comments | Same as FLAC | | WAV | RIFF INFO / ID3 | Limited support, inconsistent | ## Essential Tags ### For Music - **Title** — Song name (avoid ALL CAPS or special characters) - **Artist** — Performing artist (consistent spelling across albums) - **Album Artist** — For compilations, the album-level artist - **Album** — Album name (consistent across all tracks) - **Track Number** — Format as 'N/Total' (e.g., 3/12) - **Disc Number** — For multi-disc albums (1/2, 2/2) - **Year** — Release year (four digits: 2026) - **Genre** — Primary genre (one, not a list) - **Album Art** — Embedded JPEG, 500×500 to 1500×1500 pixels ### For Podcasts - **Title** — Episode title including episode number - **Artist** — Show name or host name - **Album** — Podcast show name - **Track Number** — Episode number - **Comment** — Episode description/show notes - **Genre** — 'Podcast' - **Album Art** — Show artwork (3000×3000 for Apple Podcasts) ## Album Art Specifications | Platform | Minimum Size | Recommended | Max File Size | |----------|-------------|-------------|---------------| | Apple Music | 1400×1400 | 3000×3000 | 10 MB | | Spotify | 640×640 | 3000×3000 | 10 MB | | Embedded (MP3) | 300×300 | 500×500 | 500 KB | ## Common Tagging Mistakes - **Inconsistent artist names** — 'The Beatles' vs 'Beatles, The' creates duplicate entries in music libraries - **Missing album artist** — Compilation tracks appear under individual artists instead of grouped together … ### Audio Compression for Streaming: Optimizing for Bandwidth Streaming audio over the internet requires balancing quality against bandwidth consumption and latency. Codec selection, bitrate adaptation, and buffering strategy determine whether listeners hear music or stuttering. - URL: https://peasyaudio.com/guides/audio-compression-streaming-optimization/ - Category: Best Practice - Reading time: 2 min - Words: 306 Key takeaways: - Modern streaming services dynamically adjust audio quality based on available bandwidth. - The optimal codec for real-time streaming. - Streaming clients maintain a playback buffer — pre-downloaded audio ahead of the current playback position. - Music service:** AAC 256 kbps default, adaptive to 128/64 on slow networks ## Bandwidth Reality | Connection | Typical Bandwidth | Supported Audio Quality | |------------|------------------|------------------------| | 2G mobile | 50-100 kbps | Low-quality speech only | | 3G mobile | 300-700 kbps | MP3 128-192 kbps | | 4G mobile | 5-50 Mbps | Any lossy, some lossless | | 5G / WiFi | 50+ Mbps | Lossless, Hi-Res | | Satellite | 1-5 Mbps | Most lossy formats | ## Adaptive Bitrate Streaming Modern streaming services dynamically adjust audio quality based on available bandwidth. Spotify offers four quality tiers (24, 96, 160, 320 kbps) and switches between them mid-stream as network conditions change. Apple Music similarly adapts between AAC 64 kbps and lossless ALAC. ## Codec Choice for Streaming ### Opus The optimal codec for real-time streaming. Opus adapts seamlessly between speech and music modes, supports bitrates from 6 kbps to 510 kbps, and has the lowest latency of any modern codec (2.5 ms minimum frame size). Used by Discord, WhatsApp, and most VoIP services. ### AAC-LC and HE-AAC AAC-LC (Low Complexity) is the standard music streaming codec. HE-AAC v2 adds spectral band replication and parametric stereo for excellent quality at very low bitrates (32-64 kbps) — used for bandwidth-constrained music streaming. ## Buffer Management Streaming clients maintain a playback buffer — pre-downloaded audio ahead of the current playback position. Typical buffer sizes: - **Music streaming:** 10-30 seconds ahead - **Live radio:** 3-5 seconds (latency trade-off) - **Video conferencing:** <200 ms (real-time requirement) Larger buffers improve reliability on unstable connections but increase initial wait time and memory usage. ## Optimizing for Your Use Case - **Music service:** AAC 256 kbps default, adaptive to 128/64 on slow networks - **Podcast delivery:** MP3 128 kbps CBR mono (widest compatibility) - **Live communication:** Opus 32-64 kbps (lowest latency) - **Background web audio:** AAC … ### Spatial Audio and 3D Sound: Formats, Creation, and Playback Spatial audio places sound in a three-dimensional space around the listener. From Dolby Atmos to binaural rendering, understanding the technology landscape helps creators choose the right approach for immersive audio experiences. - URL: https://peasyaudio.com/guides/spatial-audio-3d-sound-guide/ - Category: Comparison - Reading time: 2 min - Words: 366 Key takeaways: - Traditional stereo audio positions sounds on a left-right axis. - The dominant immersive audio format for music and cinema. - Spatial audio through headphones uses binaural rendering. - Apple Music, Amazon Music, and Tidal support Dolby Atmos streaming. ## What Spatial Audio Does Traditional stereo audio positions sounds on a left-right axis. Spatial audio adds height, depth, and rear placement — sounds can come from above, behind, and at precise locations around the listener. This creates an immersive experience where audio has the same dimensional quality as the real world. ## Spatial Audio Formats ### Dolby Atmos The dominant immersive audio format for music and cinema. Atmos uses an object-based approach: instead of assigning audio to fixed channels (left, right, center), each sound is an 'object' with a position in 3D space. The renderer adapts the mix to whatever speaker configuration or headphone system the listener uses. ### Ambisonics A mathematical representation of a 360-degree sound field. Used primarily in VR, AR, and 360-degree video. First-order Ambisonics (FOA) uses 4 channels; higher-order Ambisonics (HOA) uses 9, 16, or more channels for increased spatial resolution. ### Binaural Audio Two-channel audio processed with Head-Related Transfer Functions (HRTFs) to simulate 3D perception through headphones. The HRTF models how sound waves interact with the listener's head, ears, and torso. Works with any standard headphone — no special hardware required. ## Creation Approaches | Method | Tools | Best For | |--------|-------|----------| | Native Atmos mixing | Pro Tools, Logic Pro | Music production | | Ambisonic recording | Ambisonic microphone (Zoom H3-VR, Sennheiser AMBEO) | Field recording, VR | | Binaural recording | Dummy head microphone (Neumann KU 100) | Podcasts, ASMR | | Software spatialization | Dear Reality dearVR, Facebook 360 Spatial Workstation | Post-production | ## Playback Requirements ### Headphones Spatial audio through headphones uses binaural rendering. Head tracking (available on AirPods Pro, AirPods Max, and Sony WF-1000XM5) enhances the illusion by adjusting the sound field as the listener turns their head. ### Speakers Full speaker-based spatial audio requires a … ### Audio Restoration: Recovering Damaged and Degraded Recordings Damaged recordings — vinyl crackle, tape hiss, clipping distortion, and environmental noise — can be significantly improved with modern restoration techniques. The goal is removing damage while preserving the original character. - URL: https://peasyaudio.com/guides/audio-restoration-techniques-guide/ - Category: How-To - Reading time: 1 min - Words: 295 Key takeaways: - Process audio damage in this specific sequence: - Clipping occurs when the recording level exceeds the maximum, flattening waveform peaks into hard edges. - Aggressive noise reduction can remove more than noise — it strips away ambience, room character, and subtle musical details. - Apply each process at the minimum effective level ## Common Types of Audio Damage | Damage Type | Source | Restoration Approach | |------------|--------|---------------------| | Crackle/pops | Vinyl, damaged digital | Declicker (impulse detection) | | Hiss | Tape, analog preamps | Spectral denoiser | | Hum/buzz | Electrical interference | Notch filter (50/60 Hz + harmonics) | | Clipping | Input level too high | Declip (waveform reconstruction) | | Wow/flutter | Tape speed variation | Pitch correction | | Dropouts | Tape damage, digital errors | Interpolation | ## Restoration Order Process audio damage in this specific sequence: 1. **Declip** — Fix clipped waveforms first (they affect all subsequent processing) 2. **Declick/Decrackle** — Remove impulse noise (pops, clicks) 3. **Dehum** — Remove electrical hum and harmonics 4. **Denoise** — Reduce broadband noise (hiss, ambient) 5. **EQ correction** — Compensate for frequency response issues 6. **Final loudness adjustment** — Normalize to target level This order matters because each process works best on a signal free of the previous damage type. ## Declipping Clipping occurs when the recording level exceeds the maximum, flattening waveform peaks into hard edges. Digital clipping produces harsh, buzzy distortion. Declipping algorithms analyze the surrounding waveform shape and reconstruct the missing peaks using mathematical interpolation. ## The Over-Restoration Trap Aggressive noise reduction can remove more than noise — it strips away ambience, room character, and subtle musical details. The result sounds unnaturally clean and lifeless. A vinyl recording should still sound like vinyl, just without distracting pops and excessive hiss. ## Practical Guidelines - Apply each process at the minimum effective level - A/B compare the processed and original audio frequently - Preserve 'character' noise that defines the recording's era and medium - Keep the unprocessed original — restoration can always be redone with better tools in the future ### Audio File Size Optimization: Reducing Size Without Losing Quality Large audio files strain bandwidth, storage, and loading times. Optimizing file size requires understanding which parameters affect size and which affect perceived quality — they are not always the same. - URL: https://peasyaudio.com/guides/audio-file-size-optimization/ - Category: How-To - Reading time: 2 min - Words: 338 Key takeaways: - Audio file size is calculated as: - Music:** AAC 192-256 kbps or MP3 VBR V2 (~190 kbps) - Standard audio uses 44.1 kHz (CD) or 48 kHz (video/broadcast). - Stereo doubles the data compared to mono (two channels vs. - Audio files often contain silence at the beginning and end. ## What Determines Audio File Size Audio file size is calculated as: `Size (bytes) = Bitrate (bits/sec) × Duration (sec) / 8` A 4-minute audio file at 320 kbps: `320,000 × 240 / 8 = 9,600,000 bytes ≈ 9.6 MB` The three levers for reducing file size are bitrate, sample rate, and channel count. ## Reducing Bitrate The most effective lever. Moving from 320 kbps to 192 kbps reduces file size by 40% with minimal perceived quality loss for most content. The key is choosing the lowest bitrate that remains transparent for the content type: - **Music:** AAC 192-256 kbps or MP3 VBR V2 (~190 kbps) - **Podcasts/speech:** AAC 96-128 kbps mono (speech has less complexity) - **Background audio:** AAC 64-96 kbps (ambient, not critical listening) ## Reducing Sample Rate Standard audio uses 44.1 kHz (CD) or 48 kHz (video/broadcast). For speech-only content, 22.05 kHz captures frequencies up to 11 kHz — sufficient for voice intelligibility but cuts file size in half. Do not use reduced sample rates for music — the loss of high frequencies (cymbals, harmonics) is audible. ## Mono vs. Stereo Stereo doubles the data compared to mono (two channels vs. one). For speech content (podcasts, audiobooks, voice memos), mono is standard — there is no meaningful stereo information in a single-microphone voice recording. Converting to mono halves the file size. ## Practical Optimization Strategies | Content Type | Optimal Settings | Approx. Size (4 min) | |-------------|-----------------|--------------------| | Music (archival) | FLAC, 44.1kHz, stereo | ~25 MB | | Music (streaming) | AAC 256kbps, stereo | ~7.5 MB | | Podcast | AAC 128kbps, mono | ~3.7 MB | | Voice memo | AAC 64kbps, mono | ~1.9 MB | | Web background | AAC 64kbps, mono | ~1.9 MB | ## Silence Trimming Audio files … ### Screen Recording Best Practices: Setup, Capture, and Export High-quality screen recordings require proper setup before hitting record. Resolution, frame rate, audio, and cursor visibility all affect the final result. - URL: https://peasyaudio.com/guides/screen-recording-best-practices/ - Category: Best Practice - Reading time: 1 min - Words: 202 Key takeaways: - Record at the resolution your audience will view at. - Move the cursor deliberately and slowly. - For web delivery, export as MP4 with H.264 codec, CRF 20-23, and AAC audio at 128 kbps. - Pause briefly before clicking important elements. ## Pre-Recording Setup ### Resolution and Scaling Record at the resolution your audience will view at. For tutorials, 1080p (1920x1080) is standard. If your display is higher-res (Retina/4K), set recording to output at 1080p to avoid unnecessarily large files. ### Frame Rate Use 30fps for tutorials and presentations. Use 60fps only for gaming or UI animations where smoothness matters. Higher frame rates double file size. ### Desktop Preparation - Hide desktop icons and personal bookmarks - Close notification apps (Slack, email) - Use a clean browser profile without personal extensions - Set a neutral wallpaper (solid color) - Increase cursor size for visibility ## During Recording ### Cursor Movement Move the cursor deliberately and slowly. Viewers need time to track cursor position. Pause briefly before clicking important elements. ### Audio Use an external microphone — built-in laptop mics capture keyboard noise and fan sounds. Speak in a consistent, measured pace. Record system audio separately if needed. ## Export Settings For web delivery, export as MP4 with H.264 codec, CRF 20-23, and AAC audio at 128 kbps. This produces the best compatibility across browsers and devices. For maximum quality archival, use ProRes or H.265 at CRF 18 — you can always re-encode later. ### MIDI Basics: Understanding Digital Music Notation MIDI is not audio — it is a set of instructions that tells instruments what notes to play. Understanding this distinction unlocks powerful creative possibilities for music production, live performance, and sound design. - URL: https://peasyaudio.com/guides/midi-basics-digital-music-notation/ - Category: How-To - Reading time: 2 min - Words: 341 Key takeaways: - MIDI (Musical Instrument Digital Interface) is a communication protocol from 1983 that remains the backbone of digital music production. - Note numbers map to pitches: Middle C = 60, C#4 = 61, etc. - Music production:** Compose in MIDI, experiment with different instruments and sounds after recording - Released in 2020, MIDI 2.0 increases resolution from 7-bit (128 values) to 32-bit (4.3 billion values) for velocity, controllers, and pitch bend. ## What MIDI Is MIDI (Musical Instrument Digital Interface) is a communication protocol from 1983 that remains the backbone of digital music production. A MIDI file contains no sound — it contains instructions: 'play note C4 at velocity 80 for 500 milliseconds.' The receiving instrument (hardware synthesizer, software plugin, or virtual instrument) interprets these instructions and generates the actual sound. ## MIDI Messages ### Note Messages | Message | Data | Purpose | |---------|------|---------| | Note On | Note number (0-127), Velocity (0-127) | Start playing a note | | Note Off | Note number, Release velocity | Stop playing a note | Note numbers map to pitches: Middle C = 60, C#4 = 61, etc. Velocity represents how hard the key was pressed (0 = silent, 127 = maximum force), which typically controls volume and tone. ### Control Messages - **CC1 (Modulation):** Typically vibrato depth - **CC7 (Volume):** Channel volume - **CC10 (Pan):** Left-right stereo position - **CC64 (Sustain):** Sustain pedal on/off - **Pitch Bend:** Smoothly bends pitch up or down ## MIDI vs. Audio | MIDI | Audio (WAV/MP3) | |------|----------------| | Instructions (what to play) | Actual sound waves | | ~10 KB for a 4-min song | ~40 MB (WAV) / ~8 MB (MP3) | | Editable note by note | Editable only as waveform | | Sounds different on every instrument | Sounds identical everywhere | | No fixed timbre/tone | Fixed timbre/tone | ## Practical Uses - **Music production:** Compose in MIDI, experiment with different instruments and sounds after recording - **Live performance:** MIDI controllers (keyboards, drum pads) trigger sounds from laptops or hardware synths - **Education:** MIDI visualizations show which notes are played (popular in YouTube piano tutorials) - **Film scoring:** Compose with MIDI instruments, replace with live orchestra recordings later ## MIDI … ### Audio Effects Chain: Correct Order for Mixing and Mastering The order of audio effects in the signal chain dramatically affects the final result. Placing compression before EQ produces different results than EQ before compression. Understanding the correct sequence prevents common mixing mistakes. - URL: https://peasyaudio.com/guides/audio-effects-chain-order-guide/ - Category: Best Practice - Reading time: 2 min - Words: 327 Key takeaways: - Each effect in the chain processes the output of the previous effect. - Remove low-frequency rumble and noise before any other processing. - For mastering (processing the final stereo mix): - Reverb/Delay (Last) Time-based effects are always last in the chain. - Limiter The limiter is always the final processor, catching peaks and enforcing the loudness ceiling. ## Why Order Matters Each effect in the chain processes the output of the previous effect. A compressor placed after an EQ boost will react to the boosted frequencies more aggressively than the unboosted ones. Reversing the order produces a fundamentally different sound. There is no single 'correct' order, but there is a standard sequence that works as a starting point. ## Recommended Channel Strip Order ### 1. High-Pass Filter (First) Remove low-frequency rumble and noise before any other processing. Set the cutoff at 80-100 Hz for vocals, 30-60 Hz for instruments. This prevents subsonic energy from triggering the compressor and consuming headroom. ### 2. Subtractive EQ (Second) Cut problematic frequencies — nasal resonance (800 Hz-1 kHz), muddiness (200-400 Hz), harshness (2-4 kHz). Always cut before boosting. Subtractive EQ cleans the signal before compression, so the compressor works on a cleaner input. ### 3. Compression (Third) Even out the dynamic range. With problematic frequencies already removed by EQ, the compressor responds more predictably. Typical voice settings: 3:1 ratio, fast attack (5-10 ms), medium release (100-200 ms). ### 4. Additive EQ (Fourth) Boost desirable frequencies — vocal presence (3-5 kHz), air (10-12 kHz), warmth (200-300 Hz). Boosting after compression means the compressor does not squash the boosted frequencies. ### 5. De-Esser (Fifth) Tame sibilance (harsh 's' and 'sh' sounds) in the 5-10 kHz range. Placed after additive EQ because presence boosts can increase sibilance. ### 6. Saturation/Harmonic Enhancer (Sixth) Subtle harmonic saturation adds warmth and analog character. Apply after all corrective processing so it enhances the finished sound. ### 7. Reverb/Delay (Last) Time-based effects are always last in the chain. They process the final, polished signal and create the sense of space. Compressing after reverb would destroy the reverb's natural decay. ## Mastering Chain For mastering (processing the final stereo mix): … ### Trigonometry Basics for Developers and Designers Trigonometric functions power animations, game physics, audio visualization, and SVG path generation. This guide covers sin, cos, and tan with practical code examples. - URL: https://peasyaudio.com/guides/trigonometry-basics-for-developers/ - Category: How-To - Reading time: 2 min - Words: 304 Key takeaways: - Trigonometry is the math of angles and distances. - For a right triangle with angle θ: - Most programming languages use radians. - Place points on a circle of radius r centered at (cx, cy): ## Why Developers Need Trigonometry Trigonometry is the math of angles and distances. It underlies circular motion, wave animations, audio waveforms, radar charts, analog clocks, and collision detection. Understanding sine and cosine unlocks an enormous range of visual and interactive effects. ## The Big Three Functions For a right triangle with angle θ: - **sin(θ)** = opposite / hypotenuse - **cos(θ)** = adjacent / hypotenuse - **tan(θ)** = opposite / adjacent = sin / cos ## Radians vs Degrees Most programming languages use radians. Convert with: - Degrees to radians: `rad = deg * (π / 180)` - Radians to degrees: `deg = rad * (180 / π)` Full circle = 360° = 2π radians. ## Practical Applications ### Circular Motion Place points on a circle of radius r centered at (cx, cy): `x = cx + r * cos(angle)` `y = cy + r * sin(angle)` Distribute 12 items evenly: increment angle by 2π/12 = 30° between each. ### Wave Animation A sine wave oscillates between -1 and 1. Control amplitude (height), frequency (speed), and phase (offset): `y = amplitude * sin(frequency * x + phase)` ### Rotation Rotate point (x, y) by angle θ around the origin: `x' = x * cos(θ) - y * sin(θ)` `y' = x * sin(θ) + y * cos(θ)` ### Distance and Angle Between Points Distance: `d = sqrt((x2-x1)² + (y2-y1)²)` Angle: `θ = atan2(y2-y1, x2-x1)` Use `atan2` instead of `atan` because it handles all four quadrants correctly. ## Quick Reference | Angle | sin | cos | tan | |-------|-----|-----|-----| | 0° | 0 | 1 | 0 | | 30° | 0.5 | 0.866 | 0.577 | | 45° | 0.707 | 0.707 | 1 | | 90° | 1 | 0 | undefined | | 180° | 0 | … ### Audio Mastering Fundamentals: Preparing Mixes for Distribution Mastering is the final creative step before audio is distributed — a process of subtle refinement that ensures a mix translates well across all playback systems. Understanding the fundamentals prevents over-processing. - URL: https://peasyaudio.com/guides/audio-mastering-fundamentals-guide/ - Category: How-To - Reading time: 2 min - Words: 351 Key takeaways: - Mastering serves four purposes: - Before touching any controls, listen to the entire mix. - After mastering, check on multiple systems: studio monitors, headphones, car stereo, laptop speakers, and phone speaker. - ## Critical Listening Checks After mastering, check on multiple systems: studio monitors, headphones, car stereo, laptop speakers, and phone speaker. - Never process the low end (below 200 Hz) in the side channel — bass should remain centered. ## What Mastering Does Mastering serves four purposes: 1. **Tonal balance** — Ensure the frequency spectrum is balanced (not too bass-heavy, not too bright) 2. **Dynamic consistency** — Even out the overall dynamics so quiet and loud passages are appropriately balanced 3. **Loudness** — Bring the level to the target loudness standard for the delivery platform 4. **Translation** — Ensure the audio sounds good on every playback system: earbuds, car speakers, laptop speakers, studio monitors ## The Mastering Signal Chain ### Step 1: Reference Listening Before touching any controls, listen to the entire mix. Compare against 2-3 reference tracks in the same genre that are professionally mastered. Note specific differences: 'the low end feels thinner than the reference,' 'the vocals feel recessed compared to the reference.' ### Step 2: EQ Make broad, gentle adjustments (1-2 dB). Mastering EQ is about overall tonal balance, not surgical corrections (those belong in mixing). Common moves: high-pass at 20-30 Hz to remove subsonic content, gentle shelf boosts for 'air' above 10 kHz, gentle cuts in muddy regions (200-400 Hz). ### Step 3: Compression Glue compression with a slow attack (30+ ms), slow release (200+ ms), low ratio (1.5:1 to 2:1), and 1-3 dB of gain reduction. This subtly bonds the elements of the mix together without squashing transients. ### Step 4: Stereo Enhancement If the mix feels narrow, subtle mid-side EQ can widen the stereo image. Boost the side channel above 5 kHz for a sense of width. Never process the low end (below 200 Hz) in the side channel — bass should remain centered. ### Step 5: Limiting The final limiter sets the output ceiling (typically -1 dBTP) and brings the loudness to the target LUFS. Use 2-4 dB of limiting maximum. If more is needed, the mix is too quiet and should … ### Troubleshooting Audio Waveform Visualization Issues Audio waveform visualization — displaying the amplitude of audio over time — is used in editors, players, and web applications. Common rendering issues include silent waveforms, clipping indicators, and performance problems. - URL: https://peasyaudio.com/guides/troubleshooting-audio-waveform-visualization/ - Category: Troubleshooting - Reading time: 2 min - Words: 380 Key takeaways: - A waveform display reads audio sample data and maps amplitude values to pixel heights. - The waveform displays as a flat line even though the audio plays correctly. - The waveform appears to have flat tops and bottoms (squared-off peaks). - Loading a full-resolution waveform for a 60-minute audio file requires reading millions of samples. - The renderer reads from the appropriate peaks level based on the current zoom, never touching the raw audio data for overview rendering. ## How Waveform Rendering Works A waveform display reads audio sample data and maps amplitude values to pixel heights. For a 3-minute stereo file at 44.1 kHz, that is approximately 16 million samples. Since a display might be 800 pixels wide, each pixel represents about 20,000 samples. The renderer calculates the min and max amplitude within each pixel's sample range to draw the waveform shape. ## Common Issue: Flat or Silent Waveform ### Symptom The waveform displays as a flat line even though the audio plays correctly. ### Causes and Fixes - **Wrong channel read:** The renderer reads channel 0 (left) but the audio is only in channel 1 (right). Fix: render both channels or detect the active channel. - **Integer overflow:** Sample values are interpreted as unsigned integers instead of signed. Values cluster near zero. Fix: use the correct data type (signed 16-bit or 32-bit float). - **Byte order mismatch:** Big-endian data read as little-endian (or vice versa) produces noise-like values near zero. Fix: match the file's endianness. - **Compressed audio not decoded:** The renderer reads compressed bytes (MP3, AAC) directly instead of decoding first. Fix: decode to PCM before rendering. ## Common Issue: Clipped Waveform ### Symptom The waveform appears to have flat tops and bottoms (squared-off peaks). ### Diagnosis This may represent actual clipping in the audio (peaks exceeding 0 dBFS) or a rendering scale issue. Play the clipped section — if you hear distortion, the audio is genuinely clipped. If it sounds clean, the waveform display is not scaling correctly. ## Performance Optimization ### Problem: Slow Rendering Loading a full-resolution waveform for a 60-minute audio file requires reading millions of samples. For interactive applications (scrubbing, zooming), this is too slow. ### Solution: Pre-computed Peaks Generate a peaks data file at multiple zoom levels. Store the min/max … ### Troubleshooting Video Playback: Codec Errors, Black Screens, and Audio Sync Video playback problems — black screens, no audio, stuttering, or sync issues — usually stem from codec incompatibilities or corrupted containers. - URL: https://peasyaudio.com/guides/troubleshooting-video-playback-issues/ - Category: Troubleshooting - Reading time: 1 min - Words: 256 Key takeaways: - Cause: The video codec is not supported by the player or browser. - Cause: Mismatched container and codec. - Cause: Variable frame rate (VFR) recordings from phones or screen capture tools. - Cause: Bitrate exceeds available bandwidth or decode capability. - Cause: GPU driver issues with hardware-accelerated decoding. ## Black Screen with Audio **Cause:** The video codec is not supported by the player or browser. Common with HEVC (H.265) in Chrome/Firefox. **Fix:** Re-encode to H.264, which is universally supported. Alternatively, install the appropriate codec pack for desktop players. ## Audio but No Video (or Vice Versa) **Cause:** Mismatched container and codec. The container (MP4) may declare one codec while containing another, or the audio codec is unsupported. **Fix:** Inspect the file's actual codecs (not just the file extension). Re-encode the problematic stream only. ## Audio-Video Sync Drift **Cause:** Variable frame rate (VFR) recordings from phones or screen capture tools. Players assume constant frame rate and accumulate timing errors. **Fix:** Convert VFR to constant frame rate (CFR) by re-encoding. Set the target frame rate to match the source's average frame rate. ## Stuttering and Buffering **Cause:** Bitrate exceeds available bandwidth or decode capability. 4K HEVC requires significant CPU/GPU power. **Fix:** Reduce resolution or bitrate. Enable hardware acceleration in the player. For streaming, use adaptive bitrate with a lower starting quality. ## Green or Purple Artifacts **Cause:** GPU driver issues with hardware-accelerated decoding. Common with AMD GPUs and VP9/AV1 content. **Fix:** Update GPU drivers. Disable hardware acceleration in the browser (Settings > System). If the issue is in an exported file, the encoder may have a bug — try a different encoder. ## File Won't Open **Cause:** Corrupted container header, incomplete download, or wrong file extension. **Fix:** Try renaming to the correct extension. Use a repair tool that rebuilds the container index from the stream data. ## Glossary (44 terms) ### Aliasing (Audio) (Audio Aliasing) Distortion occurring when audio frequencies above the Nyquist limit are sampled, producing false lower frequencies. - URL: https://peasyaudio.com/glossary/aliasing-audio/ ### Audio Codec (Audio Codec) Software that encodes raw audio into a compressed format and decodes it back for playback (e.g. MP3, AAC, FLAC). - URL: https://peasyaudio.com/glossary/audio-codec/ ### Audio Fingerprint (Acoustic Fingerprinting) A compact digital summary of an audio signal used to identify songs or detect duplicate content. - URL: https://peasyaudio.com/glossary/audio-fingerprint/ ### Audio Watermark (Audio Watermarking) Embedding imperceptible identifying data into an audio signal for copyright tracking and content identification. - URL: https://peasyaudio.com/glossary/audio-watermark/ ### Bit Depth (Audio) (Audio Bit Depth) The number of bits in each audio sample, determining the dynamic range and noise floor (e.g. 16-bit, 24-bit). - URL: https://peasyaudio.com/glossary/bit-depth-audio/ ### Bitrate (Audio Bitrate) The amount of audio data processed per second, measured in kbps, directly affecting quality and file size. - URL: https://peasyaudio.com/glossary/bitrate/ ### Channel (Audio Channel) An independent audio signal in a recording (mono = 1, stereo = 2, surround = 5.1+). - URL: https://peasyaudio.com/glossary/channel/ ### Chorus Effect (Audio Chorus Effect) An effect that duplicates a signal with slight pitch and timing variations to simulate multiple voices or instruments. - URL: https://peasyaudio.com/glossary/chorus-effect/ ### Clipping (Audio Clipping) Distortion that occurs when an audio signal exceeds the maximum level a system can handle. - URL: https://peasyaudio.com/glossary/clipping/ ### Codec (Audio/Video Codec) Software that encodes and decodes audio or video data, determining the compression algorithm used. - URL: https://peasyaudio.com/glossary/codec/ ### Compression (Audio) (Dynamic Range Compression) Reducing the volume difference between the loudest and quietest parts of an audio signal. - URL: https://peasyaudio.com/glossary/compression-audio/ ### Crossfade (Audio Crossfade) A smooth transition between two audio clips where one fades out while the other fades in simultaneously. - URL: https://peasyaudio.com/glossary/crossfade/ ### De-esser (Audio De-esser) A specialized compressor targeting sibilant frequencies (4-10 kHz) to reduce harsh 's' and 'sh' sounds in vocals. - URL: https://peasyaudio.com/glossary/de-esser/ ### Delay (Audio Delay Effect) An effect that records and plays back audio after a set time interval, creating echo and rhythmic repetitions. - URL: https://peasyaudio.com/glossary/delay/ ### Dithering (Audio) (Audio Dithering) Adding low-level random noise when reducing bit depth to mask quantization distortion and preserve detail. - URL: https://peasyaudio.com/glossary/dithering-audio/ ### Dynamic Range (Audio Dynamic Range) The difference between the quietest and loudest parts of an audio signal, measured in decibels. - URL: https://peasyaudio.com/glossary/dynamic-range/ ### Equalization (Audio Equalization) The process of adjusting the balance of frequency components in an audio signal. - URL: https://peasyaudio.com/glossary/equalization/ ### Fade (Audio Fade) A gradual increase (fade in) or decrease (fade out) in audio volume over time. - URL: https://peasyaudio.com/glossary/fade/ ### FLAC (Free Lossless Audio Codec) An open-source audio codec that compresses audio without any loss in quality, reducing file sizes by 50-70%. - URL: https://peasyaudio.com/glossary/flac/ ### Frequency Spectrum (Audio Frequency Spectrum) The range of frequencies present in an audio signal, typically from 20 Hz (bass) to 20 kHz (treble) for human hearing. - URL: https://peasyaudio.com/glossary/frequency-spectrum/ ### Harmonic Distortion (Total Harmonic Distortion) The presence of frequencies that are integer multiples of the fundamental, sometimes added intentionally for warmth. - URL: https://peasyaudio.com/glossary/harmonic-distortion/ ### Limiter (Audio Limiter) A compressor with a very high ratio that prevents audio from exceeding a set maximum level. - URL: https://peasyaudio.com/glossary/limiter/ ### Loudness War (Loudness War) The trend of increasing audio loudness in mastering at the expense of dynamic range, countered by LUFS standards. - URL: https://peasyaudio.com/glossary/loudness-war/ ### LUFS (Loudness Units Full Scale) A standardized loudness measurement reflecting perceived volume, used for broadcast and streaming normalization targets. - URL: https://peasyaudio.com/glossary/lufs/ ### Mid-Side Processing (Mid-Side Audio Processing) A technique separating stereo audio into center (mid) and sides signals for independent processing. - URL: https://peasyaudio.com/glossary/mid-side-processing/ ### Mono (Monaural Audio) Audio reproduced through a single channel, where all sound comes from one source. - URL: https://peasyaudio.com/glossary/mono/ ### Noise Gate (Audio Noise Gate) A processor that silences audio below a threshold level, eliminating background noise between sounds. - URL: https://peasyaudio.com/glossary/noise-gate/ ### Noise Reduction (Audio Noise Reduction) Techniques for removing unwanted background noise from audio recordings. - URL: https://peasyaudio.com/glossary/noise-reduction/ ### Normalization (Audio Normalization) The process of adjusting audio volume to a target level without changing the dynamic range. - URL: https://peasyaudio.com/glossary/normalization/ ### Nyquist Frequency (Nyquist Frequency) Half the sample rate, representing the highest frequency that can be accurately captured in digital audio. - URL: https://peasyaudio.com/glossary/nyquist-frequency/ ### PCM (Pulse-Code Modulation) The standard method of digitally representing analog audio signals by sampling amplitude at regular intervals. - URL: https://peasyaudio.com/glossary/pcm/ ### Peak Level (Audio Peak Level) The highest instantaneous amplitude in an audio signal, used to prevent clipping and set headroom. - URL: https://peasyaudio.com/glossary/peak-level/ ### Pitch (Audio Pitch) The perceived frequency of a sound, determining how high or low a tone sounds to the listener. - URL: https://peasyaudio.com/glossary/pitch/ ### Psychoacoustics (Psychoacoustics) The study of how humans perceive sound, informing audio compression, spatial audio, and loudness standards. - URL: https://peasyaudio.com/glossary/psychoacoustics/ ### Reverb (Reverberation) The persistence of sound in a space after the original sound stops, simulated digitally for spatial effects. - URL: https://peasyaudio.com/glossary/reverb/ ### RMS Level (Root Mean Square Level) An average audio level measurement that better represents perceived loudness than peak level readings. - URL: https://peasyaudio.com/glossary/rms-level/ ### Sample Rate (Audio Sample Rate) The number of audio samples captured per second, measured in Hz (e.g. 44,100 Hz for CD quality). - URL: https://peasyaudio.com/glossary/sample-rate/ ### Sidechain (Sidechain Processing) Using one audio signal to control the processing of another, commonly used to duck music under voiceover. - URL: https://peasyaudio.com/glossary/sidechain/ ### Spatial Audio (Spatial Audio) 3D sound technology that positions audio sources in virtual space around the listener using HRTF and head tracking. - URL: https://peasyaudio.com/glossary/spatial-audio/ ### Spectrogram (Audio Spectrogram) A visual representation of audio frequencies over time, showing the spectrum of a signal. - URL: https://peasyaudio.com/glossary/spectrogram/ ### Stereo (Stereophonic Audio) Audio using two independent channels (left and right) to create a sense of spatial width. - URL: https://peasyaudio.com/glossary/stereo/ ### Stereo Imaging (Stereo Imaging) Techniques for widening, narrowing, or repositioning the perceived stereo field of an audio mix. - URL: https://peasyaudio.com/glossary/stereo-imaging/ ### Tempo (Audio Tempo) The speed or pace of an audio track, measured in beats per minute (BPM). - URL: https://peasyaudio.com/glossary/tempo/ ### Waveform (Audio Waveform) A visual representation of an audio signal showing amplitude over time. - URL: https://peasyaudio.com/glossary/waveform/ ## API Base URL: https://peasyaudio.com/api/v1/ | Endpoint | Description | |----------|-------------| | GET /api/v1/tools/ | List all tools (filterable by category) | | GET /api/v1/tools/{slug}/ | Tool detail with description, steps, FAQ | | GET /api/v1/categories/ | Tool categories | | GET /api/v1/formats/ | File formats reference | | GET /api/v1/formats/{slug}/ | Format detail with history, pros/cons | | GET /api/v1/conversions/ | Format conversion guides | | GET /api/v1/glossary/ | Terminology with definitions | | GET /api/v1/glossary/{slug}/ | Term with simple + technical explanations, code examples, references | | GET /api/v1/guides/ | Educational guides (filterable by category, audience_level) | | GET /api/v1/guides/{slug}/ | Full guide with content, takeaways, ToC, related guides | | GET /api/v1/use-cases/ | Real-world use cases by industry | | GET /api/v1/search/?q={query} | Cross-model search | | GET /api/v1/sites/ | All 16 Peasy sites | OpenAPI spec: https://peasyaudio.com/api/openapi.json ## Content Negotiation All pages support Markdown output for AI agents: - Add `?format=md` to any page URL - Or send `Accept: text/markdown` header ## Platform Stats - 251 tools across 16 sites - 131 file formats documented - 1484 format conversion guides - 695 glossary terms (with simple + technical explanations) - 645 educational guides (avg 1,308 words) - 302 real-world use cases