What Is PCM Audio? The Format Behind WAV and AIFF
PCM Is Not a File Format — It's a Method
Most people encounter PCM audio through file formats like WAV or AIFF, which leads to a common misconception: that PCM is itself a container. It isn't. PCM stands for Pulse-Code Modulation, and it describes the process by which analog audio signals are converted into digital data. WAV and AIFF are simply wrappers — file containers — that most commonly store PCM-encoded audio inside them. The PCM process works in two steps. First, the analog waveform is sampled at regular intervals. Second, each sample's amplitude is quantized — assigned a numerical value within a fixed range. The result is a stream of numbers that, when played back in sequence, reconstruct the original sound wave with high fidelity. This is fundamentally different from compressed audio formats like MP3 or AAC, where mathematical models discard audio data deemed less perceptible to human hearing. PCM keeps everything. A 10-second recording at CD quality (44,100 Hz sample rate, 16-bit depth, stereo) produces exactly 10 × 44,100 × 2 bytes × 2 channels = roughly 1.76 MB of raw audio data, every time, with no shortcuts taken. Understanding this distinction matters practically. When a video editor says 'I need uncompressed audio,' they mean PCM. When a mastering engineer insists on 'lossless source files,' they mean PCM. The format is the foundation of professional audio work, even if the name rarely appears on file icons or export menus.
Sample Rate and Bit Depth: The Two Numbers That Define PCM Quality
Every PCM audio file is characterized by two core parameters: sample rate and bit depth. These numbers appear constantly in audio software, and knowing what they actually mean prevents costly mistakes during recording, editing, and conversion. Sample rate is measured in Hertz (Hz) and describes how many times per second the analog signal is measured. The Nyquist theorem establishes that a sample rate must be at least twice the highest frequency you want to capture. Human hearing tops out around 20,000 Hz, which is why the CD standard of 44,100 Hz was chosen — it provides a comfortable margin above 40,000 Hz. Common sample rates you'll encounter are 44,100 Hz (CD, streaming), 48,000 Hz (video production and broadcast), 88,200 Hz and 96,000 Hz (high-resolution audio), and 192,000 Hz (studio archival work). Choosing 48,000 Hz for a podcast is not wrong, but it does mean your file won't align natively with CD-spec assets without resampling. Bit depth determines how many distinct amplitude values each sample can hold. At 16-bit, each sample can be one of 65,536 possible values. At 24-bit, that jumps to 16,777,216 values. The practical consequence is dynamic range: 16-bit PCM delivers approximately 96 dB of dynamic range, while 24-bit provides around 144 dB. Recording at 24-bit gives engineers headroom to capture quiet passages without pushing gain too high, then reduce to 16-bit at the final delivery stage. A common error is converting a 48,000 Hz / 24-bit file down to 44,100 Hz / 16-bit carelessly. The sample rate conversion (SRC) algorithm matters enormously here — poor SRC introduces aliasing artifacts. Professional tools like iZotope RX or Adobe Audition use high-quality SRC by default. CocoConvert applies standard SRC algorithms for common conversion paths, which is sufficient for most web and podcast use cases, but if you're preparing a commercial music master, dedicated mastering software is the appropriate tool.
WAV vs. AIFF: Two Containers, One Codec
WAV (Waveform Audio File Format) was developed by Microsoft and IBM and introduced with Windows 3.1 in 1991. AIFF (Audio Interchange File Format) was created by Apple in 1988, based on IFF from Electronic Arts. Both formats were designed to store PCM audio without compression, and for the vast majority of practical purposes, a 44,100 Hz / 16-bit stereo WAV and an equivalent AIFF are sonically identical — the audio data inside is the same PCM stream. The differences are structural and ecosystem-related. WAV files use little-endian byte ordering (the least significant byte comes first), while AIFF uses big-endian ordering. This was relevant when PowerPC Macs and x86 PCs used different processor architectures, but modern software handles both transparently. WAV files support metadata through LIST-INFO chunks and the more capable BWF (Broadcast Wave Format) extension used in film and TV production, which embeds timecode and scene data. AIFF supports metadata through its MARK and NAME chunks, and the AIFF-C variant supports compressed audio encoding, though this is rarely used. In practice: Windows audio software defaults to WAV, Apple's Logic Pro defaults to AIFF, and video production tools like Premiere Pro and DaVinci Resolve accept both equally. If you're delivering audio to a client and don't know their platform, WAV is the safer universal choice. Neither format has a quality advantage over the other when both contain linear PCM data at the same sample rate and bit depth.
Where PCM Audio Actually Lives in Your Workflow
PCM audio appears at more points in a typical media workflow than most people realize. Understanding where it sits helps you make better decisions about when to convert and when to leave files alone. In video production, the audio tracks embedded in professional video formats — MXF, ProRes, and DNxHD — almost always carry PCM audio at 48,000 Hz / 24-bit. When a video editor exports a final cut from Premiere Pro (File > Export > Media, then setting Audio Format to AAC or PCM), they're deciding at that moment whether to preserve the PCM or compress it. Broadcast delivery specifications like those from the BBC (R/68) or Netflix explicitly require PCM audio in certain deliverable formats. In music production, DAWs like Ableton Live, Logic Pro, and Pro Tools record and process audio internally as PCM. Ableton Live 11 records at the project sample rate (set in Preferences > Audio) in 32-bit float WAV by default — a PCM variant that uses floating-point numbers instead of integers, giving engineers more headroom during mixing without clipping. The 32-bit float files are then typically converted to 16-bit or 24-bit PCM for delivery. For archival purposes, libraries, broadcasters, and archives store audio as PCM WAV or AIFF precisely because there is no codec to become obsolete. An MP3 encoder from 2001 may produce files that sound different from a 2024 encoder at the same bitrate. A PCM file from 1991 decoded today is bit-for-bit identical to how it was stored. The Library of Congress recommends PCM WAV as a preferred format for audio preservation for exactly this reason. Consumer use cases are simpler: ripping a CD in iTunes or Music.app with 'AIFF Encoder' selected gives you PCM AIFF files. Ripping with Windows Media Player to WAV gives you PCM WAV. Both are lossless copies of the disc.
Converting PCM Audio: What Changes and What Doesn't
Converting between PCM formats — WAV to AIFF, or vice versa — is a container swap. The audio data doesn't change, only the file wrapper. This is a lossless operation regardless of the tool you use, including CocoConvert. You can convert a WAV to AIFF and back to WAV indefinitely without any degradation. Converting from PCM to a compressed format like MP3, AAC, or OGG Vorbis is a different matter entirely. This is a lossy operation. The encoder analyzes the PCM data and discards information based on psychoacoustic models. At high bitrates — 320 kbps MP3, 256 kbps AAC — the difference is inaudible to most listeners on most playback systems. At lower bitrates — 128 kbps or below — artifacts become audible, particularly on transient-heavy material like percussion and acoustic guitar. This loss is permanent and irreversible. Converting an MP3 back to WAV does not recover the discarded data; you get a large file that still sounds like an MP3. Converting from compressed audio back to PCM is something CocoConvert can do technically — an MP3 to WAV conversion is a valid operation and the resulting WAV file will play correctly. But it's important to be transparent: the audio quality ceiling is set by the source MP3, not the WAV container. A 128 kbps MP3 converted to WAV is a 128 kbps MP3's worth of audio in a much larger file. The only legitimate reason to do this conversion is software compatibility — some older hardware or software accepts WAV but not MP3. For quality recovery, it's not useful. When converting PCM to PCM at different sample rates or bit depths, quality depends on the algorithm. Downsampling from 96,000 Hz to 44,100 Hz requires a low-pass filter to prevent aliasing, and the quality of that filter varies between tools. For critical listening material, dedicated audio software with high-quality SRC is preferable.
PCM Variants You'll Encounter: Float, LPCM, and DPCM
Standard integer PCM is not the only form of PCM you'll encounter, and the variants matter in specific contexts. 32-bit float PCM (also called IEEE 754 float) stores sample values as floating-point numbers rather than integers. This format is standard in DAW internal processing because it handles values outside the normal 0–1 range without hard clipping, making it more forgiving during mixing. Ableton Live, Pro Tools, and FL Studio all use 32-bit float internally. The files are larger than 24-bit integer PCM (4 bytes per sample vs. 3 bytes), and most consumer playback software handles them correctly, but some older hardware may not. When delivering finished audio, you'll typically dither down to 24-bit or 16-bit integer PCM. LPCM (Linear PCM) is simply PCM where the quantization steps are uniform in size — each step between amplitude values is equal. This is what all the formats discussed above use. The 'linear' qualifier distinguishes it from logarithmic PCM variants used in telephony, like A-law and μ-law (mu-law) encoding, which compress the dynamic range of speech to fit into 8-bit samples efficiently. You'll encounter A-law and μ-law in telephony files with extensions like .au or in certain WAV files from phone systems. CocoConvert can convert these to standard linear PCM WAV, which is often necessary before editing in standard audio software. DPCM (Differential PCM) and its adaptive variant ADPCM store the difference between successive samples rather than absolute values. ADPCM appears in some game audio formats and older multimedia containers. It's a form of compression, and while it's technically PCM-derived, it's not lossless. IMA ADPCM WAV files, for example, have a characteristic slightly degraded sound compared to linear PCM at the same sample rate and bit depth.
Choosing the Right PCM Settings for Common Use Cases
Selecting the correct PCM settings isn't about chasing the highest numbers — it's about matching the requirements of your destination platform and preserving quality through the production chain. For music distribution to streaming platforms (Spotify, Apple Music, Tidal): Deliver 44,100 Hz / 16-bit stereo WAV for standard quality, or 44,100 Hz / 24-bit for platforms that accept hi-res uploads. Spotify's internal encoding pipeline accepts up to 44,100 Hz / 16-bit; delivering 96,000 Hz doesn't improve what listeners hear through their encoder. Apple Music's Lossless tier accepts 44,100 Hz or 48,000 Hz at 24-bit. For video production and broadcast: Use 48,000 Hz / 24-bit PCM consistently. This is the broadcast standard worldwide. Mixing at 48,000 Hz avoids sample rate conversion when the audio is married to video, which almost universally runs at 48,000 Hz. For podcasting and voice content: 44,100 Hz / 16-bit WAV is more than sufficient for recording and editing. Most podcast hosting platforms re-encode to MP3 or AAC anyway, so delivering lossless PCM at the highest reasonable quality gives the encoder the best source material. There's no benefit to recording a podcast at 192,000 Hz. For archival and preservation: 96,000 Hz / 24-bit PCM WAV is a reasonable standard that captures more than enough information and remains practical in terms of storage. The Library of Congress and most national archives specify this range. To convert your PCM audio files — WAV to AIFF, adjust sample rate, or convert to compressed formats for delivery — CocoConvert handles the most common conversion paths directly in your browser without requiring software installation. For specialized tasks like high-quality SRC for mastering, or batch processing hundreds of broadcast files with BWF metadata preservation, dedicated tools like iZotope RX or Adobe Audition are the appropriate choice. Knowing the limits of any tool, including this one, is part of working with audio professionally.