CBRConstant bitrate. The encoder aims to keep the stream at the selected fixed rate.
ABRAverage bitrate. LAME targets an overall average while allowing individual frames to vary.
VBRVariable bitrate. Quality is prioritised and bitrate changes with the complexity of the audio.
cVBR / constrained VBRVariation is limited more tightly than normal VBR. CW extensions may add stricter minimum-rate behaviour.
MP3 · Lossy
LAME 4.0 cVBRb
Performance / Solid State
MP3 is still one of the most widely supported compressed audio formats. The supplied LAME 4.0 cVBRb builds are intended for users who want MP3 compatibility while retaining more control over the lower end of a variable-bitrate encode. Performance is the normal choice for modern systems; Solid State is the compatibility-focused alternative.
Codec typeLossy MP3
Standard LAME modesCBR · ABR · VBR
cVBRb extensionStrict minimum bitrate + bitrate boost
Current EAC preset-V0 · strict min 192 kbps · boost 3
-V0 -b 192 --vbr-min-strict --bitrate-boost 3
The supplied cVBRb builds extend the normal LAME 4.0 feature set rather than replacing it. Standard LAME CBR, ABR and VBR operation remains available, while cVBRb adds stricter lower-bitrate control to the constrained-VBR path.
For MP3, fixed CBR rates use the normal Layer III bitrate set, while ABR chooses an average target and VBR is quality-driven through the usual -V scale. The EAC preset shown here uses -V0 together with a strict 192 kbps minimum and bitrate boost.
Because cVBRb is an extension of standard LAME 4.0, the site describes its additional controls separately from LAME's established CBR/ABR/VBR behaviour.
Why this build?Performance uses the SIMD/multithreaded Visual Studio path; Solid State prioritises compatibility. Both retain standard LAME 4.0 operation and the cVBRb controls.
Open the EAC Codecs LAME guide →
FLAC · Lossless
FLAC 1.5.0
Official FLAC tools
FLAC compresses PCM audio without discarding audio information. Decoding returns the lossless audio, making FLAC well suited to CD archiving and general high-quality playback. Unlike a lossy codec, FLAC does not use a target audio bitrate to determine quality.
Codec typeLossless
Bitrate modeNot target-bitrate based
Current EAC settingCompression level 5
Audio qualityLossless at every compression level
Compression level changes encoding effort and resulting file size rather than decoded audio quality. The verified EAC command on this site uses level 5.
Open the FLAC EAC guide →
AAC · Lossy · M4A
CW-AAC 1.0.1.0
FFmpeg AAC implementation
AAC is a modern perceptual codec commonly carried in an M4A/MP4 container. It is useful where broad modern-device support and efficient lossy compression are more important than MP3's near-universal legacy compatibility.
Documented modesVBR · pVBR
Recommended EAC modeVBR quality 2
Alternative documented modepVBR quality 2
EAC Bit rate dropdownNot used / command-line controlled
CW-AAC expects the input and output as its first two positional arguments. The recommended EAC setup therefore keeps %source% %dest% at the start of the command line.
Why this build?CW-AAC provides direct M4A metadata handling plus documented VBR and pVBR workflows for EAC.
Open the AAC EAC guide →
ALAC · Lossless · M4A
CW-ALAC 1.0.1.0
Apple Lossless
ALAC is a lossless codec stored in an M4A container. It is particularly useful when lossless storage is required alongside strong compatibility with Apple software and devices.
Codec typeLossless
ModeALAC lossless encoding
Target bitrateNot applicable
MetadataWritten directly into M4A
There is no lossy quality or bitrate mode to select: output bitrate varies with the source audio while preserving lossless reconstruction.
Why this build?CW-ALAC writes lossless M4A output and metadata directly, including common tags and artwork support.
Open the ALAC EAC guide →
Opus · Lossy
CW-Opus 1.6.1.0
libopus based
Opus is a modern, high-efficiency lossy codec with strong software playback support. CW-Opus automatically resamples input when required for an Opus-compatible sample rate.
Rate-control modesVBR · constrained VBR · protective cVBR · CBR
Default bitrate128 kbps
Current EAC preset256 kbps protective cVBR
Complexity0–10 · default 10
Frontend sample rates8 / 12 / 16 / 24 / 44.1 / 48 kHz
Frame sizes2.5 / 5 / 10 / 20 / 40 / 60 ms
ChannelsMono · Stereo
EAC Bit rate dropdownNot used / command-line controlled
VBR is the normal/default Opus operating mode. -cvbr by itself selects standard constrained VBR, while -cvbr together with an explicit -b value enables CW protective cVBR, where the requested bitrate acts as a minimum for eligible CELT frames. -cbr selects constant bitrate.
The default requested bitrate is 128 kbps. The EAC guide deliberately uses -cvbr -b 256000 --complexity 10, giving a 256 kbps protective-cVBR configuration with maximum encoder complexity.
44.1 kHz input is automatically resampled to 48 kHz by the CW 64-tap Blackman-windowed sinc frontend. Native Opus API sample rates bypass that resampler.
Why this build?CW-Opus adds protective cVBR controls and handles 44.1 kHz EAC input through its dedicated 64-tap sinc frontend.
Open the Opus EAC guide →
AC-3 · Lossy surround
CW-AC3 1.0.1.0
Dolby Digital compatible
AC-3 is a lossy multichannel format commonly associated with Dolby Digital playback. For the EAC workflow, standard 44.1 kHz CD PCM is automatically resampled to 48 kHz before AC-3 encoding.
Encoding mode48 kHz CBR
Legal bitrates19 rates · 32–640 kbps
Accepted input rates44.1 / 48 / 88.2 / 96 / 176.4 / 192 kHz
ChannelsStereo through multichannel · 7.1 → 5.1 rear fold
CW-AC3 produces 48 kHz AC-3 using constant bitrate encoding at all 19 legal AC-3 bitrates from 32 through 640 kbps. In the normal CD workflow, 44.1 kHz EAC input is therefore converted to 48 kHz automatically.
The encoder also provides matrix-stereo decoding with a 20 ms default rear delay (adjustable from 0–40 ms), and a --thx preset that enables fixed 80 Hz LR4 bass management.
Why this build?CW-AC3 accepts CD-rate input, covers all 19 legal AC-3 bitrates, and includes matrix/THX processing options.
Open the AC-3 EAC guide →
DTS Core · Lossy surround
CW-DTS 1.0.1.0
Native CD-rate support
DTS Core is a lossy multichannel format used by a wide range of home-cinema and optical-disc playback systems. CW-DTS is especially relevant to EAC because it supports native 44.1 kHz CD-rate PCM rather than forcing the EAC source to 48 kHz.
Public CBR profiles768 / 1024 / 1152 / 1411.2 kbps
Public sample rates44.1 / 48 kHz
Public layouts2.0 / 3.0 / 4.0 / 5.0 / 5.1
EAC Bit rate dropdownNot used / command-line controlled
CW-DTS exposes four constrained DTS Core bitrate profiles: 768, 1024, 1152 and 1411.2 kbps. The command-line syntax uses 1411 for the highest profile, mapping exactly to 1,411,200 bit/s.
Public input support covers 44.1 and 48 kHz PCM, including native CD-rate 44.1 kHz operation. The encoder supports stereo and validated 3.0, 4.0, 5.0 and 5.1 layouts. Optional matrix processing shares the CW-AC3 steering behaviour, including a 20 ms default rear delay and the --thx 80 Hz LR4 bass-management preset.
Why this build?CW-DTS supports native 44.1 kHz DTS Core, four public CBR profiles and validated multichannel layouts through 5.1.
Open the DTS EAC guide →
Specialist CD audio tools
DTS-CD · Specialist tool
CW-DTS-WAV 1.0.1.0
Extract / decode
DTS-CD places a compressed DTS bitstream inside audio-CD sample words. A rip can therefore look like ordinary 44.1 kHz, 16-bit stereo PCM even though the payload contains DTS data. The carrier must remain bit-perfect.
Input carrier44.1 kHz · 16-bit · stereo WAV
ModesScan · raw DTS extraction · decode
Decoded output24-bit multichannel WAV / FLAC
Target bitrateNot applicable
Raw .dts output is a bitstream extraction rather than a decode/re-encode operation. FLAC output is decoded lossless PCM and preserves the channel layout reported by the DTS decoder.
Keep DTS-CD rips bit-perfect. Do not normalise, alter volume, apply DSP, resample, fade or perform any other processing that changes the carrier PCM values.
Open the DTS-CD EAC guide →
HDCD · Specialist decoder
CW-HDCD 1.0.1.0
Decode / preserve
CW-HDCD detects and reconstructs supported HDCD signalling carried in stereo PCM audio. Normal decoding produces 24-bit PCM without resampling, normalisation, ReplayGain, additional DSP or dithering.
Main modesScan · decode · Preserve-HDCD
Decode input16-bit stereo WAV/RF64 or FLAC
Decode output24-bit stereo WAV/RF64 or FLAC
Target bitrateNot applicable
Preserve-HDCD performs lossless PCM16 container conversion without reconstructing the HDCD signal, preserving the original PCM sample values and embedded signalling unchanged.
Open the HDCD EAC guide →