AAC

The MPEG-4 successor to MP3 — better sound at similar bitrates, with three available variants.

Pro Audio Converter can convert audio files to and from AAC (.m4a, .aac).

Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio. Designed as the MPEG-4 successor to MP3, AAC generally achieves better sound quality than MP3 at similar bitrates. AAC is also the default or standard audio format for iPhone, iPod, iPad, and Apple Music. Pro Audio Converter offers the following AAC variants:

AAC — Produces the most common AAC LC (Low Complexity) audio format. This is the format you get when purchasing or exporting music from Apple Music or iTunes or QuickTime.

MPEG 4 High Efficiency AAC — AAC HE (High Efficiency) is an extension of Low Complexity AAC (AAC LC), optimized for low-bitrate applications such as streaming audio. AAC-HE streams are compatible with AAC-LC decoders.

MPEG 4 Low Delay AAC — AAC LD (Low Delay) is optimized for playback latency and is often used for real-time applications like telephony or conferencing (for example, iChat).

Encoding Options

Sample Rate — The number of samples of audio carried per second, measured in Hz or kHz (1000 Hz). 44.1 kHz is the sampling rate of audio CDs and 48.0 kHz is commonly used for professional video. Higher sample rates result in higher quality audio with larger file sizes. Setting this to Auto creates an output file with the same sample rate as the input file. If the output file does not support the sample rate of the input file, Pro Audio Converter uses the greatest sample rate that is supported.

Bitrate Mode

Constant Bitrate (CBR) — Uses the same number of bits for every frame. Whether the passage is difficult or easy to encode, the encoder uses the same number of bits, so AAC quality is variable; complex parts are of lower quality than easy ones. The advantage is that the final file size won't change and can be accurately predicted.

Variable Bit Rate (VBR) — You choose the desired quality on a scale from 1 (lowest quality / highest distortion) to 10 (highest quality / lowest distortion). The encoder tries to maintain the given quality throughout by choosing the optimal number of bits for each part of the music. You control quality directly, but file size is unpredictable.

Average Bit Rate (ABR) — You choose a target bitrate, and the encoder tries to maintain that as an average while using higher bitrates for passages that need more bits. Higher quality than CBR at a predictable average file size. This is the default and recommended mode for encoding AAC.

Variable Bit Rate Constrained — Similar to VBR but limits the average bit rate variation. The lower limit is the user-selected bit rate; the higher bit rate adapts for difficult tracks and can produce larger files than ABR. Recommended as a compromise between VBR and ABR.

Bitrate — The number of binary values used to encode each second of audio. 128 kbps (kilobits per second) = 128,000 bits per second. This bitrate is shared across all simultaneous audio channels. Guide values:

<64 kbps — average to low quality. Some audible degradation. Acceptable for non-critical streaming.
64–96 kbps — good quality. No noticeable compression artefacts. Fine for most vocal uses.
128–160 kbps — high quality. Transparent to most listeners. Suitable for music and sensitive audio.
160 kbps and up — indistinguishable from the original to most listeners.

Channels — The number of audio channels in the output file. Auto attempts to match the input file's channel count. You can also force Stereo (2 channels) or Mono (1 channel).

Quality — Good is optimized for the fastest encoding. Choose Better or Best (optimal for 24-bit sources) for higher quality. The tradeoff is between encoding speed and audio quality.