Convert WAV to AMR

Uncompressed audio container format developed by Microsoft and IBM in 1991. WAV stores raw PCM samples, preserving every detail of the original recording with zero quality loss. The de facto standard for professional audio production, recording, and mastering on Windows and cross-platform DAWs.

A narrow-band speech codec standardized by 3GPP in 1999, designed primarily for mobile voice communication. AMR operates at 8 kHz sampling rate with variable bitrates from 4.75 to 12.2 kbps, dynamically adapting to network conditions. Widely used by Android and Nokia phones for voice memos and call recordings, AMR delivers intelligible speech in extremely small file sizes.

WAV stores raw PCM samples - each audio sample is written directly without compression or transformation:

# Convert to WAV (16-bit, 44.1 kHz)
ffmpeg -i input.mp3 -codec:a pcm_s16le \
  -ar 44100 output.wav

# High-resolution WAV (24-bit, 48 kHz)
ffmpeg -i input.flac -codec:a pcm_s24le \
  -ar 48000 output.wav

AMR uses Algebraic Code-Excited Linear Prediction (ACELP) to model speech signals, encoding 20 ms frames at variable bitrates:

# Encode audio to AMR at default bitrate
ffmpeg -i input.wav -ar 8000 -ac 1 \
  -codec:a libopencore_amrnb output.amr

# Specify bitrate mode (12.2 kbps best)
ffmpeg -i input.wav -ar 8000 -ac 1 \
  -b:a 12.2k output.amr

• Metadata: INFO/LIST chunks, BWF metadata
• Album Art: Not natively supported
• Gapless Playback: Inherent - no encoder padding
• Streaming: Poor - large file sizes
• Surround: Multichannel PCM up to 18 channels
• Chapters: Supported via cue chunks

• Metadata: Minimal - no standard tagging system
• Album Art: Not supported
• Gapless Playback: Not applicable (speech codec)
• Streaming: Excellent for mobile networks (low bandwidth)
• Surround: Not supported (mono only)
• Adaptive Rate: Dynamic bitrate switching per 20 ms frame

• Bit-perfect audio reproduction with zero quality loss
• Industry standard for recording, editing, and mastering
• Compatible with every DAW and audio editor
• Supports high-resolution audio (24-bit/192 kHz)
• No generation loss when re-editing or re-saving
• Multichannel support for surround sound

• Extremely small file sizes (under 1 MB for several minutes of speech)
• Optimized for human voice with high intelligibility
• Dynamic bitrate adaptation to network conditions
• Native support on virtually all mobile phones
• Low CPU requirements for encoding and decoding
• 3GPP standard ensures broad telecom compatibility

• Very large files (~10 MB/min at CD quality)
• Impractical for streaming or mobile storage
• No built-in compression option
• Limited native metadata support
• 4 GB file size limit (RIFF limitation)

• 8 kHz narrow-band - poor quality for music
• Mono only - no stereo or surround support
• Maximum 12.2 kbps bitrate severely limits fidelity
• Limited metadata and tagging capabilities
• Not suitable for any content beyond speech

• Studio recording and multitrack sessions
• Audio editing and post-production
• Mastering and final mix rendering
• Broadcast and radio playout systems
• Sound design and sample libraries
• CD authoring and disc burning

• Mobile phone voice memos and recordings
• Voicemail storage on cellular networks
• MMS audio attachments
• Telecom voice logging and archival
• Low-bandwidth voice transmission

• Professional audio editing and mixing in a DAW
• Archiving master recordings at full quality
• Creating source files for encoding to other formats
• Broadcast production with strict quality standards

• Recording voice notes on Android devices
• Storing large volumes of speech recordings compactly
• Mobile voice communication applications
• Embedded systems with limited storage