Configuration help
Generic
Title
Name of the renderer, displayed in control points and other devices that support displaying name of the device.
Backend
Select one of the support audio output backends. Either ALSA or Network Audio (NAA).
Output mode
Select either fixed output mode, either PCM or SDM, or alternatively [source] to automatically switch between PCM and SDM depending on source material.
Fixed volume
When not enabled, volume can be adjusted from remote control points. When set to fixed, volume control is not available to remote control points and the volume is set to specified fixed value.
Max volume
Sets maximum level of volume control range.
Min volume
Sets minimum level of volume control range.
Startup volume
When volume control is variable, this specifies the default volume setting at startup time.
PCM gain compensation
Due to nature of DSD, many DACs have different output levels for 0 dBFS PCM vs 0 dB DSD. PCM gain compensation can be used to compensate for this level difference.
| DAC type | Compensation (dB) |
|---|---|
| Asahi Kasei Micro (AKM), AK4490 | -3.5 |
| Asahi Kasei Micro (AKM), AK4493 | -1 to -3.5 depending on reference level settings |
| Asahi Kasei Micro (AKM), AK4499 | -4.1 |
| Asahi Kasei Micro (AKM), AK4499EX | -3 depending on settings |
| Cirrus Logic | -3 |
| ESS Sabre | 0 |
| Texas Instruments / Burr-Brown | Depends on selected AFIR, refer to the datasheet for details. |
| Holo Audio | -6 |
| Denafrips | -3.2 |
| Merging Hapi | -0.6 |
Adaptive volume
Support for ReplayGain 2.0 to adjust volume offset based on loudness metadata. For album transport, album gain is used. For playlist transport, track gain is used. It is also important to note that some of the gain metadata may contain positive gain values, which need to be taken into account on the volume setting in order to avoid triggering limiter.
Note! To leave headroom for inter- sample overs in reconstruction, it is not recommended to set volume higher than -3 dBFS. Pay attention to "Limits" counter on front page, it should stay 0 at all times! There is no reason to go for "loudness wars" way and try to squeeze every last dB out of the DAC.
Channels
Specify number of output channels to use.
FFT filter length
This option specifies length of the FFT filter, per each 2x cascade. Default value is 512. Thus the length is conversion rate invariant. Length affects steepness of the filter, shorter lengths result in slower (gentler) roll-off, while higher lengths result in faster (steeper) roll-off.
Idle time
Adjust amount of time engine is allowed to run idling before stopping, once playback has ended. Restarting playback from idle is faster than from stopped state.
DSP pipelines
Number of DSP pipelines to allocate. Minimum required number of DSP pipelines is maximum number of input or output channels. Or number of matrix pipelines in use.
Options
Pre before meter performs pre-process
functions beforing metering. By default these
are performed after metering. This allows
one to see effect of for example 20 kHz filter
in Client, but it also makes harder to
recognize when the filter should be turned
off.
Auto rate family switches output
sampling rate based on source material such
way that both source and output rate have
a common base rate. For example 44.1 kHz or
48 kHz.
Quick pause uses simple silence pattern
for faster pause response, but increased
likelihood of audible glitches when pause is
toggled.
Short buffer adjusts FIFO buffer length
allowing to balance between playback delay
and safety against drop-outs. Shorter the FIFO
buffer, more likely it is to experience drop-
outs. Short option uses half length FIFO
buffer for audio for faster control responses,
but increased risk of drop-outs. Minimal option
keeps minimum possible amount of data in FIFO
buffer, but this also makes overall system
sensitive to drop-outs.
UPnP
Freewheel immediately fetches entire track to RAM at full speed, when possible. This works only for tracks of known size and cannot work for "endless" realtime streams. May also cause issues with gapless playback due to the sudden traffic spike.
Log file
Log output can be enabled for troubleshooting purposes and the location can be specified.
DSD sources
This group controls settings that may be relevant when source format it DSD. Depending on cases described below.
Direct SDM
When Direct SDM is enabled and DAC supports DSD input, DSD sources go through bit-perfect when output mode is SDM. When Direct SDM is disabled and output mode is SDM, Integrator and SDM-SDM Conversion parameters apply. When output mode is PCM, Gain +6 dB, SDM-PCM Conversion and Noise filter parameters apply. Note! When Direct SDM is enabled, volume control is set to fixed -3.5 dB value also for PCM sources when output mode is set to SDM to avoid sudden jumps in volume at format changing track transition.
Integrator
Selects integrator structure type for the SDM remodulation loop.
| Integrator | Description |
|---|---|
| IIR | Normal IIR type |
| IIR2 | IIR type integrator structure designed to minimize residual noise. |
| IIR3 | High order IIR type integrator structure. |
| FIR | Weighted FIR type |
| FIR2 | Weighted FIR type integrator structure. 50 kHz audio bandwidth re DSD64. |
| FIR-bl | FIR type integrator structure with band-limiting. 24 kHz audio bandwidth re DSD64 with complete cut by 45 kHz. |
| FIR-bw | FIR type integrator structure with brickwall band-limiting. 21.5 kHz audio bandwidth re DSD64 with complete cut by 30 kHz. |
| CIC | Cascade comb type |
SDM-SDM Conversion
Selects rate conversion algorithm type for SDM-to-SDM rate conversions. Difference between different algorithms is in ultrasonic frequency response.
| Rate converter | Description |
|---|---|
| wide | Default wide bandwidth |
| narrow | Alternative medium bandwidth |
| XFi | Extreme fidelity |
Gain +6 dB
For DSD, 0 dB reference level is defined to be 6 dB below 100% modulation index. Specification allows exceeding this reference level up to about +3 dB for short periods. This setting applies +6 dB gain for DSD sources bringing the typical subjective level in line with PCM sources. This increases possibility for limiting, which can be countered same way as for PCM by using HQPlayer's volume control.
SDM-PCM Conversion
Rate down-conversion algorithm used for SDM-to-PCM conversions. Output PCM sampling rate of the conversion is 1/16th of the original SDM rate.
| Conversion type | Description |
|---|---|
| traditional | Traditional recursive conversion algorithm. Minimizes amount of ringing by using slow roll-off filters. |
| single-steep | Single-pass with steep cut-off. |
| single-short | Single-pass with normal cut-off. Optimized tradeoff between ringing and wide frequency response. |
| sinc-S | Adaptive length single-pass with very steep cut-off. (4096 x ratio taps) |
| sinc-M | Million tap single-pass with very steep cut-off. |
| poly-lp | Linear-phase single-pass. |
| poly-mp | Minimum-phase single-pass. |
| poly-short-lp | Linear-phase slow roll-off single-pass. |
| poly-short-mp | Minimum-phase slow roll-off single-pass. |
| poly-xtr | Linear-phase extreme cut-off and attenuation single-pass. |
| poly-xtr-short | Short linear-phase extreme cut-off and attenuation single-pass. |
| poly-ext2 | Linear-phase with sharp roll-off and very high attenuation single-pass. |
| poly-gauss-long | Very steep and high attenuation single-pass Gaussian. |
| none | No rate conversion. Mostly suitable when output is very high rate PCM. |
Noise filter
Filter for removing shaped noise of DSD material.
| Noise filter | Description |
|---|---|
| standard | Standard noise filter (DSD spec) will be applied. |
| low | Similar to standard, but has lower corner frequency and results in almost flat noise profile in ultrasonic range. |
| high-order | High order noise filter designed for material created with high order modulators. |
| sac | Moving average converter combined with fourth order IIR, simulating behavior of the Signalyst DSC-1 DAC. |
| wec | Weighted element converter. |
| wec2 | Weighted element converter. Optimized to closely match DSD/SACD specification. |
| slow-lp | Slow roll-off linear-phase filter. |
| slow-mp | Slow roll-off minimum-phase filter. |
| medium | Medium roll-off linear-phase filter designed to be as gentle as possible while passing minimal amount of out-of-band noise. |
| medium-high | Medium roll-off high reate linear-phase filter designed to be as gentle as possible while passing minimal amount of out-of-band noise. |
| fast-lp | Fast roll-off linear-phase filter. |
| fast-mp | Fast roll-off minimum-phase filter. |
| brickwall | Brickwall filter that doesn’t pass any out-of-band noise. Very steep linear phase filter. Cut-off at 25 kHz for DSD64, 50 kHz for DSD128, 100 kHz for DSD256, 200 kHz for DSD512 and 400 kHz for DSD1024. |
PCM settings
These settings apply only to PCM output mode.
1x Filter
Rate conversion filter applied when the source sampling rate is ≤ 50 kHz, so called "1x rate".
Nx Filter
Rate conversion filter applied when the source sampling rate is > 50 kHz, so called "2x rate" and higher.
| Filter | Description | Special focus | Genre | Ratio | Apodizing |
|---|---|---|---|---|---|
| none | No sample rate conversion happens. Only sample depth (word length) is changed as needed. | 1:1 | N | ||
| IIR | This is analog-sounding filter, especially suitable for recordings containing strong transients, long post-echo is a side effect (not usually audible due to masking). A really steep IIR filter is used. This filter type is similar to analog filters and has no pre-echo, but has a long post-echo. Small amount of pass-band ripple is also present. Medium attenuation. IIR filter is applied in time-domain. | Pop, rock, jazz, blues | Integer | Y | |
| IIR2 | This is analog-sounding filter, especially suitable for recordings containing strong transients, long post-echo is a side effect (not usually audible due to masking). A steep IIR filter is used. This filter type is similar to analog filters and has no pre-echo, but has a long post-echo. Medium attenuation. IIR filter is applied in time-domain. | Pop, rock, jazz, blues | Integer | Y | |
| FIR | Typical “oversampling” digital filter, generally suitable for most uses (slight pre- and post-echo), but best on classical music recorded in a real world acoustic environment such as concert hall. This is the most ordinary filter type, usually present in hardware. This filter is applied in time-domain. It has average amount of pre- and post-echo. | Classical | Integer | Y | |
| asymFIR | Asymmetric FIR, good for jazz/blues, and other music containing transients recorded in real world acoustic environment. Otherwise same as FIR, but with a shorter pre-echo and longer post-echo. Modifies phase response, but not as much as minimum phase FIR. | Jazz, blues | Integer | Y | |
| minphaseFIR | Minimum phase FIR, good for pop/rock/electronic music containing strong transients such as drums and percussion and where recording is made in a studio using multi-track equipment. No pre-echo, but somewhat long post-echo. | Pop, rock, electronic | Integer | Y | |
| FFT | Technically good steep “brickwall” filter, but might have some side effects (pre-echo) on material containing strong transients. This filter is similar to FIR, but it is applied in frequency-domain and is quite efficient from performance point of view while having rather long impulse response. Length can be configured separately with the FFT filter length option. | Classical | 2x | Y | |
| poly-sinc-lp better space |
Linear phase polyphase sinc filter. Very high quality linear phase resampling filter, can perform most of the typical conversion ratios. Good phase response, but has some amount of pre-echo. See “FIR” for further details. | Space | Classical | Any | Y |
| poly-sinc-mp better transients |
Minimum phase polyphase sinc filter, otherwise similar to poly-sinc. Altered phase response, but no pre-echo. See “minphaseFIR” for further details. | Transients | Jazz, blues | Any | Y |
| poly-sinc-short-lp | Otherwise similar as poly-sinc, but shorter pre- and post-echos at the expense of filtering quality (slower roll-off). | Space, transients | Jazz, blues, electronic | Any | Y |
| poly-sinc-short-mp | Minimum phase variant of poly-sinc-short. Otherwise similar to poly-sinc-mp, but shorter post-echo (slower roll-off). Most optimal transient reproduction. | Transients | Pop, rock | Any | Y |
| poly-sinc-long-lp | Otherwise similar as
poly-sinc, but longer pre- and
post-echos with improved
filtering quality (faster
roll-off). Warning! Initialization of this filter for 44.1k to 48k family conversion can take huge amount of time! |
Space | Classical | Any | Y |
| poly-sinc-long-ip | Intermediate phase
version of poly-sinc-long,
with small pre-echo and longer
post-echo with improved
filtering quality (faster
roll-off). Warning! Initialization of this filter for 44.1k to 48k family conversion can take huge amount of time! |
Space, transients | Jazz, blues, electronic | Any | Y |
| poly-sinc-long-mp | Minimum phase variant of
poly-sinc-long. Otherwise
similar to poly-sinc-mp, but
longer post-echo with improved
filtering quality (faster
roll-off). Warning! Initialization of this filter for 44.1k to 48k family conversion can take huge amount of time! |
Transients | Pop, rock | Any | Y |
| poly-sinc-hb | Linear-phase polyphase half-band filter with steep roll-off and high attenuation. Only suitable for highest technical quality source materials. | Any | N | ||
| poly-sinc-hb-xs | Extremely short linear-phase polyphase half-band filter with slow roll-off and low attenuation. Only suitable for highest technical quality source materials. | Pop, rock | Any | N | |
| poly-sinc-hb-s | Short linear-phase polyphase half-band filter with slow roll-off and average attenuation. Only suitable for highest technical quality source materials. | Pop, rock | Any | N | |
| poly-sinc-hb-m | Medium linear-phase polyphase half-band filter with average roll-off and medium attenuation. Only suitable for highest technical quality source materials. | Any | Any | N | |
| poly-sinc-hb-l | Long linear-phase polyphase half-band filter with fast roll-off and high attenuation. Only suitable for highest technical quality source materials. | Classical, jazz, blues | Any | N | |
| poly-sinc-ext | Linear phase polyphase sinc filter with sharper roll-off and somewhat lower stop-band attenuation, while being roughly equal length to poly-sinc. | Integer up | Y | ||
| poly-sinc-ext2 | Linear phase polyphase sinc filter with sharp roll-off and high stop-band attenuation for extended frequency response while completely cutting off by Nyquist frequency (non-halfband). | Timbre | Any | Any | Y |
| poly-sinc-ext3 | Very steep 8 times longer version of poly-sinc-ext2. | Timbre | Classical | Any | Y |
| poly-sinc-mqa/mp3-lp | Linear phase polyphase sinc filter. At 1x source rates optimized for playing MQA encoded content without decoding, or MP3 content, in order to clean up high frequency anomalies added by the encoding process. At Nx source rates suitable for rendering decoded MQA content and upsampling 88.2 kHz or higher source sampling rate, especially for hires PCM recordings of 176.4 kHz or higher sampling rate. Very short ringing. Early slow roll-off. | Transients | Classical, jazz, blues | Integer up | Y |
| poly-sinc-mqa/mp3-mp | Minimum phase variant of poly-sinc-mqa. | Transients | Pop, rock | Integer up | Y |
| poly-sinc-xtr-lp | Linear phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Classical | Any | N |
| poly-sinc-xtr-mp | Minimum phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Jazz, blues | Any | N |
| poly-sinc-xtr-short-lp | Short linear phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Electronic, jazz, blues, pop, rock | Any | Y |
| poly-sinc-xtr-short-mp | Short minimum phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Pop, rock | Any | Y |
| poly-sinc-gauss-short | Short Gaussian polyphase sinc filter. Optimal time- frequency response. | Transients | Electronic, jazz, blues, pop, rock | Integer up | Y |
| poly-sinc-gauss | Gaussian polyphase sinc filter. Optimal time- frequency response. | Transients, timbre | Any | Any | Y |
| poly-sinc-gauss-long | Long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time-frequency response. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-xl | Extra long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time-frequency response. | Transients, timbre, space | Classical, jazz, blues | Any | N |
| poly-sinc-gauss-xla | Apodizing extra long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time- frequency response. | Transients, timbre, space | Classical, jazz, blues | Any | Y |
| poly-sinc-gauss-hires-lp | Linear-phase Gaussian filter for HiRes content with extremely high attenuation. Optimal time-frequency response. Also suitable for playback of lossy compression such as MP3 or MQA. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-hires-ip | Intermediate-phase Gaussian filter for HiRes content with extremely high attenuation. Optimal time-frequency response. Also suitable for playback of lossy compression such as MP3 or MQA. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-hires-mp | Minimum-phase Gaussian filter for HiRes content with extremely high attenuation. Optimal time-frequency response. Also suitable for playback of lossy compression such as MP3 or MQA. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-halfband | Linear-phase halfband Gaussian filter. Slightly leaky around Nyquist, but extremely high attenuation. Only suitable for highest quality source materials. | Transients, timbre, space | Any | Any | N |
| poly-sinc-gauss-halfband-s | Short linear-phase halfband Gaussian filter. Leaky around Nyquist, but high attenuation. Only suitable for highest quality source materials. | Transients, timbre, space | Any | Any | N |
| ASRC | This is a special type of filter, slightly similar to FIR, but with a possibility of asynchronous operation for conversions from any rate to any other (on the fly variable) rate. Computationally heavy and not recommended. | Any | N | ||
| polynomial-1 | Polynomial interpolation. No apparent pre- or post-ringing. Frequency response rolls off slowly in the top octave. Poor stop-band rejection and will thus leak fairly high amount of ultrasonic distortion. These type of filters are sometimes referred to as “non-ringing” by some manufacturers. Not recommended. | Integer up | N | ||
| polynomial-2 | Similar to polynomial-1, but higher stop-band rejection and only one cycle of pre- and post-ringing. Not recommended. | Integer up | N | ||
| minringFIR-lp | Minimum ringing FIR. Uses special algorithm to create a linear-phase filter that minimizes amount of ringing while providing better frequency-response and attenuation than polynomial interpolators. Performance and ringing is between polynomial and poly-sinc-short. | Transients | Integer up | N | |
| minringFIR-mp | Minimum phase variant of minringFIR. | Transients | Integer up | N | |
| closed-form | Closed form interpolation with high number of taps. | 2x up | N | ||
| closed-form-fast | Closed form interpolation with lower CPU load, but also lower precision. Output precision tuned to match about 24-bit PCM. | 2x up | N | ||
| closed-form-M | Closed form interpolation with one million taps. | 2x up | N | ||
| sinc-S | sinc-filter with adaptive number of taps (4096 x ratio). Very sharp roll-off and high attenuation. | Space, timbre | Any | Integer | Y |
| sinc-M | sinc-filter with one million taps. Very sharp roll-off and high attenuation. | Space, timbre | Classical, jazz, blues | Integer | Y |
| sinc-Mx | Constant time version of sinc-M filter. Filter length is constant in time, with million taps at 16x PCM output rates. | Space, timbre | Classical, jazz, blues | Integer | Y |
| sinc-MG | Gaussian constant time filter with million taps at 16x PCM output rates. Extremely high attenuation. | Transients, timbre, space | Classical, jazz, blues | Integer | N |
| sinc-MGa | Apodizing Gaussian constant time filter with million taps at 16x PCM output rates. Extremely high attenuation. | Transients, timbre, space | Classical, jazz, blues | Integer | Y |
| sinc-L | sinc-filter with adaptive number of taps (131070 x conversion ratio). Extremely sharp roll-off but average attenuation. | Classical | Integer | N | |
| sinc-Ls | Average attenuation sinc-filter with adaptive number of taps. (4096 x ratio) | Any | Integer | N | |
| sinc-Lm | Average attenuation sinc-filter with adaptive number of taps. (16384 x ratio) | Classical, jazz, blues | Integer | N | |
| sinc-Ll | Average attenuation sinc-filter with adaptive number of taps. (65536 x ratio) | Classical | Integer | N | |
| sinc-short | Short average attenuation sinc-filter with adaptive number of taps. | Any | Any | N | |
| sinc-medium | Average attenuation sinc-filter with adaptive number of taps. | Classical, jazz, blues | Any | N | |
| sinc-long | Long average attenuation sinc-filter with adaptive number of taps. | Classical | Any | N |
Dither
Dither and noise-shaper used for producing samples at output sample depth (word length). Since internal processing always happens at much higher resolution than output allows, resolution needs to be limited to the actual output format in a proper way. Dithers and noise-shapers randomize quantization error (rounding or truncation) of the limited output resolution avoiding quantization distortion.
| Dither / NS | Description |
|---|---|
| none | No noise-shaping or dithering, only rounding. Only useful for bit-perfect playback when filter is set to none, volume is fixed to 0 dBFS and no other processing is used. Not recommended. |
| NS1 | Simple first order noise-shaping. Sample values are rounded and the quantization error is shaped such way that the error energy is pushed to the higher frequencies. Suitable mostly for ≥ 352.8 kHz upsampling. |
| NS4 | Fourth order noise-shaping. Similar in shape as “shaped” dither. Suitable for rates ≥ 88.2 kHz. |
| NS5 | Fifth order noise-shaping. Fairly aggressive noise-shaping designed for 8x and 16x rates (352.8/384/705.6/768 kHz). Not recommended for rates ≤ 192 kHz. (Especially good for BB PCM1704 DAC chip and other R2R DACs at those highest rates.) |
| NS9 | Ninth order noise-shaping. Very aggressive noise-shaping designed especially for 4x rates (176.4/192 kHz) and recommended for these rates. (Especially good for older 16-bit 4x rate capable R2R DAC chips like TDA154x etc.) |
| LNS15 | 15th order linear noise-shaping. Smooth noise shaping curve designed specifically for 16x rates (705.6/768 kHz). Can be also used at 8x rates. |
| RPDF | Rectangular Probability Density Function. White noise dither. Computationally light weight, but only suitable for 24-bit or higher output hardware. |
| TPDF | Triangular Probability Density Function. This is the industry standard simple dither mechanism. Suitable for any rate and recommended if playback rate is 44.1/48 kHz. Recommended for general purpose use. |
| Gauss1 | Gaussian Probability Density Function. High quality flat frequency dither recommended for rates at or below 96 kHz where noise-shaping is not suitable. |
| shaped | Shaped dither. Noise used in this dither has shaped frequency distribution to lower audibility of the dither noise. Suitable for playback rates ≥ 88.2 kHz. |
Sample rate
Fixed output sample rate to use, can be used to lock output to a single optimal sampling rate. When set to Auto, output rate is highest allowed by DAC capabilities, Rate limit setting and capabilities of the selected filter.
Rate limit
Default preferred output rate and maximum rate for automatic selection.
SDM settings
These settings apply only to SDM output mode.
1x Oversampling
Oversampling filter applied for PCM sources when the source sampling rate is ≤ 50 kHz, so called "1x rate".
Nx Oversampling
Oversampling filter applied for PCM sources when the source sampling rate is > 50 kHz, so called "2x rate" and higher.
| Filter | Description | Special focus | Genre | Ratio | Apodizing |
|---|---|---|---|---|---|
| IIR | This is analog-sounding filter, especially suitable for recordings containing strong transients, long post-echo is a side effect (not usually audible due to masking). A really steep IIR filter is used. This filter type is similar to analog filters and has no pre-echo, but has a long post-echo. Small amount of pass-band ripple is also present. Medium attenuation. IIR filter is applied in time-domain. | Pop, rock, jazz, blues | Integer | Y | |
| IIR2 | This is analog-sounding filter, especially suitable for recordings containing strong transients, long post-echo is a side effect (not usually audible due to masking). A steep IIR filter is used. This filter type is similar to analog filters and has no pre-echo, but has a long post-echo. Medium attenuation. IIR filter is applied in time-domain. | Pop, rock, jazz, blues | Integer | Y | |
| FIR | Typical “oversampling” digital filter, generally suitable for most uses (slight pre- and post-echo), but best on classical music recorded in a real world acoustic environment such as concert hall. This is the most ordinary filter type, usually present in hardware. This filter is applied in time-domain. It has average amount of pre- and post-echo. | Classical | Integer | Y | |
| asymFIR | Asymmetric FIR, good for jazz/blues, and other music containing transients recorded in real world acoustic environment. Otherwise same as FIR, but with a shorter pre-echo and longer post-echo. Modifies phase response, but not as much as minimum phase FIR. | Jazz, blues | Integer | Y | |
| minphaseFIR | Minimum phase FIR, good for pop/rock/electronic music containing strong transients such as drums and percussion and where recording is made in a studio using multi-track equipment. No pre-echo, but somewhat long post-echo. | Pop, rock, electronic | Integer | Y | |
| FFT | Technically good steep “brickwall” filter, but might have some side effects (pre-echo) on material containing strong transients. This filter is similar to FIR, but it is applied in frequency-domain and is quite efficient from performance point of view while having rather long impulse response. Length can be configured separately with the FFT filter length option. | Classical | 2x | Y | |
| poly-sinc-lp better space |
Linear phase polyphase sinc filter. Very high quality linear phase resampling filter, can perform most of the typical conversion ratios. Good phase response, but has some amount of pre-echo. See “FIR” PCM filter for further details. | Space | Classical | Any | Y |
| poly-sinc-mp better transients |
Minimum phase polyphase sinc filter, otherwise similar to poly-sinc. Altered phase response, but no pre-echo. See “minphaseFIR” PCM filter for further details. | Transients | Jazz, blues | Any | Y |
| poly-sinc-short-lp | Otherwise similar as poly-sinc, but shorter pre- and post-echos at the expense of filtering quality (slower roll-off). | Space, transients | Jazz, blues, electronic | Any | Y |
| poly-sinc-short-mp | Minimum phase variant of poly-sinc-short. Otherwise similar to poly-sinc-mp, but shorter post-echo (slower roll-off). Most optimal transient reproduction. | Transients | Pop, rock | Any | Y |
| poly-sinc-long-lp | Otherwise similar as poly-sinc, but longer pre- and post-echos with improved filtering quality (faster roll-off). | Space | Classical | Any | Y |
| poly-sinc-long-ip | Intermediate phase version of poly-sinc-long, with small pre-echo and longer post-echo with improved filtering quality (faster roll-off). | Space, transients | Jazz, blues, electronic | Any | Y |
| poly-sinc-long-mp | Minimum phase variant of poly-sinc-long. Otherwise similar to poly-sinc-mp, but longer post-echo with improved filtering quality (faster roll-off). | Transients | Pop, rock | Any | Y |
| poly-sinc-hb | Linear-phase polyphase half-band filter with steep roll-off and high attenuation. | Any | N | ||
| poly-sinc-hb-xs | Extremely short linear-phase polyphase half-band filter with slow roll-off and low attenuation. Only suitable for highest technical quality source materials. | Pop, rock | Any | N | |
| poly-sinc-hb-s | Short linear-phase polyphase half-band filter with slow roll-off and average attenuation. Only suitable for highest technical quality source materials. | Pop, rock | Any | N | |
| poly-sinc-hb-m | Medium linear-phase polyphase half-band filter with average roll-off and medium attenuation. Only suitable for highest technical quality source materials. | Any | Any | N | |
| poly-sinc-hb-l | Long linear-phase polyphase half-band filter with fast roll-off and high attenuation. Only suitable for highest technical quality source materials. | Classical, jazz, blues | Any | N | |
| poly-sinc-ext | Linear phase polyphase sinc filter with sharper roll-off and somewhat lower stop-band attenuation, while being roughly equal length to poly-sinc. | Integer | Y | ||
| poly-sinc-ext2 | Linear phase polyphase sinc filter with sharp roll-off and high stop-band attenuation for extended frequency response while completely cutting off by Nyquist frequency (non-halfband). Processing is two stages with minimum factor of 16 before applying special second stage. If difference between source and output rates is less than 32x, operates as a single stage with only the first stage. | Timbre | Any | Any | Y |
| poly-sinc-ext3 | Very steep 8 times longer version of poly-sinc-ext2. | Timbre | Classical | Any | Y |
| poly-sinc-mqa-lp | Linear phase polyphase sinc filter. At 1x source rates optimized for playing MQA encoded content without decoding in order to clean up high frequency noise added by the MQA encoding. At Nx source rates suitable for rendering decoded MQA content and upsampling 88.2 kHz or higher source sampling rate, especially for hires PCM recordings of 176.4 kHz or higher sampling rate. Very short ringing. | Transients | Classical, jazz, blues | Any | Y |
| poly-sinc-mqa-mp | Minimum phase variant of poly-sinc-mqa. | Transients | Pop, rock | Any | Y |
| poly-sinc-xtr-lp | Linear phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Classical | Any | N |
| poly-sinc-xtr-mp | Minimum phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Jazz, blues | Any | N |
| poly-sinc-xtr-short-lp | Short linear phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Electronic, jazz, blues, pop, rock | Any | Y |
| poly-sinc-xtr-short-mp | Short minimum phase polyphase sinc filter with extreme roll-off and attenuation. | Timbre | Pop, rock | Any | Y |
| poly-sinc-gauss-short | Short Gaussian polyphase sinc filter. Optimal time-frequency response. | Transients | Electronic, jazz, blues, pop, rock | Integer | Y |
| poly-sinc-gauss | Gaussian polyphase sinc filter. Optimal time-frequency response. | Transients, timbre | Any | Any | Y |
| poly-sinc-gauss-long | Long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time-frequency response. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-xl | Extra long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time-frequency response. | Transients, timbre, space | Classical, jazz, blues | Any | N |
| poly-sinc-gauss-xla | Apodizing extra long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time- frequency response. | Transients, timbre, space | Classical, jazz, blues | Any | Y |
| poly-sinc-gauss-hires-lp | Linear-phase Gaussian filter for HiRes content with extremely high attenuation. Optimal time-frequency response. Also suitable for playback of lossy compression such as MP3 or MQA. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-hires-ip | Intermediate-phase Gaussian filter for HiRes content with extremely high attenuation. Optimal time-frequency response. Also suitable for playback of lossy compression such as MP3 or MQA. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-hires-mp | Minimum-phase Gaussian filter for HiRes content with extremely high attenuation. Optimal time-frequency response. Also suitable for playback of lossy compression such as MP3 or MQA. | Transients, timbre, space | Any | Any | Y |
| poly-sinc-gauss-halfband | Linear-phase halfband Gaussian filter. Slightly leaky around Nyquist, but extremely high attenuation. Only suitable for highest quality source materials. | Transients, timbre, space | Any | Any | N |
| poly-sinc-gauss-halfband-s | Short linear-phase halfband Gaussian filter. Leaky around Nyquist, but high attenuation. Only suitable for highest quality source materials. | Transients, timbre, space | Any | Any | N |
| poly-sinc-*-2s | Two stage oversampling. First stage rate conversion is performed by at least by factor of 8 using the selected algorithm. And further converted to the final rate using algorithm optimized for conversion of content that has already been processed to at least 8x the source rate. If difference between source and output rates is less than 16x, operates as single stage with only the first stage. This lowers the overall CPU load, while preserving the same conversion quality. Especially useful for highest output rates. | Any | |||
| polynomial-1 | Polynomial interpolation. No apparent pre- or post-ringing. Frequency response rolls off slowly in the top octave. Poor stop-band rejection and will thus leak fairly high amount of ultrasonic distortion. These type of filters are sometimes referred to as “non-ringing” by some manufacturers. Not recommended. | Integer up | N | ||
| polynomial-2 | Similar to polynomial-1, but higher stop-band rejection and only one cycle of pre- and post-ringing. Not recommended. | Integer up | N | ||
| minringFIR-lp | Minimum ringing FIR. Uses special algorithm to create a linear-phase filter that minimizes amount of ringing while providing better frequency-response and attenuation than polynomial interpolators. Performance and ringing is between polynomial and poly-sinc-short. | Transients | Integer | N | |
| minringFIR-mp | Minimum phase variant of minringFIR. | Transients | Integer | N | |
| closed-form | Closed form interpolation with high number of taps. | 2x | N | ||
| closed-form-fast | Closed form interpolation with lower CPU load, but also lower precision. Output precision tuned to match about 24-bit PCM. | 2x | N | ||
| closed-form-16M | Closed form interpolation with 16 million taps. | 2x | N | ||
| sinc-S | sinc-filter with adaptive number of taps (4096 x ratio). Very sharp roll-off and high attenuation. | Space, timbre | Any | Integer | Y |
| sinc-M | sinc-filter with one million taps. Very sharp roll-off and high attenuation. | Space, timbre | Classical, jazz, blues | Integer | Y |
| sinc-Mx | Constant time version of sinc-M filter. Filter length is constant in time, with 4 million taps at DSD64 output rates. | Space, timbre | Classical, jazz, blues | Integer | Y |
| sinc-MG | Gaussian constant time filter with 4 million taps at DSD64 output rates. Extremely high attenuation. | Transients, timbre, space | Classical, jazz, blues | Integer | N |
| sinc-MGa | Apodizing Gaussian constant time filter with 4 million taps at DSD64 output rates. Extremely high attenuation. | Transients, timbre, space | Classical, jazz, blues | Integer | Y |
| sinc-L | sinc-filter with adaptive number of taps (131070 x conversion ratio). Extremely sharp roll-off but average attenuation. | Classical | Integer | N | |
| sinc-Ls | Average attenuation sinc-filter with adaptive number of taps. (4096 x ratio) | Any | Integer | N | |
| sinc-Lm | Average attenuation sinc-filter with adaptive number of taps. (16384 x ratio) | Classical, jazz, blues | Integer | N | |
| sinc-Ll | Average attenuation sinc-filter with adaptive number of taps. (65536 x ratio) | Classical | Integer | N | |
| sinc-short | Short average attenuation sinc-filter with adaptive number of taps. Two stage with 16x intermediate rate. | Any | Any | N | |
| sinc-medium | Average attenuation sinc-filter with adaptive number of taps. Two stage with 16x intermediate rate. | Classical, jazz, blues | Any | N | |
| sinc-long | Long average attenuation sinc-filter with adaptive number of taps. Two stage with 16x intermediate rate. | Classical | Any | N |
Modulator
Noise-shaping modulator to produce high rate output bit streams.
| Modulator | Description |
|---|---|
| DSD5 | Rate adaptive fifth order one-bit delta-sigma modulator. |
| DSD5v2 | Revised fifth order fixed configuration one-bit delta-sigma modulator. |
| DSD5v2 256+fs | Revised fifth order one-bit delta-sigma modulator optimized for rates ≥ 10 MHz. |
| DSD5EC | Rate adaptive fifth order one-bit delta-sigma modulator with extended compensation. |
| ASDM5 | Adaptive fifth order one-bit delta-sigma modulator. |
| ASDM5EC | Adaptive fifth order one-bit delta-sigma modulator with extended compensation. |
| ASDM5ECv2 | Second generation of ASDM5EC with minor improvements. |
| ASDM5ECv3 | Third generation of ASDM5EC with minor improvements. |
| ASDM5EC-ul | Adaptive fifth order one-bit delta-sigma modulator with extended compensation. Ultralight version. |
| ASDM5EC-light | Adaptive fifth order one-bit delta-sigma modulator with extended compensation. Light version. |
| ASDM5EC-super | Adaptive fifth order one-bit delta-sigma modulator with extended compensation. Super version. |
| ASDM5EC-ul 512+fs | Adaptive fifth order one-bit delta-sigma modulator with extended compensation. Optimized for 512x and higher rates. Ultralight version. |
| ASDM5EC-light 512+fs | Adaptive fifth order one-bit delta-sigma modulator with extended compensation. Optimized for 512x and higher rates. Light version. |
| ASDM5EC-super 512+fs | Adaptive fifth order one-bit delta-sigma modulatro with extended compensation. Optimized for 512x and higher rates. Super version. |
| DSD7 | Seventh order fixed configuration one-bit delta-sigma modulator. |
| DSD7 256+fs | Seventh order one-bit delta-sigma modulator optimized for rates ≥ 10 MHz. |
| ASDM7 | Adaptive seventh order one-bit delta-sigma modulator. |
| ASDM7EC | Adaptive seventh order one-bit delta-sigma modulator with extended compensation. |
| ASDM7ECv2 | Second generation of ASDM7EC with minor improvements. |
| ASDM7ECv3 | Third generation of ASDM7EC with minor improvements. |
| ASDM7EC-ul | Adaptive seventh order one-bit delta-sigma modulator with extended compensation. Ultralight version. |
| ASDM7EC-light | Adaptive seventh order one-bit delta-sigma modulator with extended compensation. Light version. |
| ASDM7EC-super | Adaptive seventh order one-bit delta-sigma modulator with extended compensation. Super version. |
| ASDM7EC-ul 512+fs | Adaptive seventh order one-bit delta-sigma modulator with extended compensation. Optimized for 512x and higher rates. Ultralight version. |
| ASDM7EC-light 512+fs | Adaptive seventh order one-bit delta-sigma modulator with extended compensation. Optimized for 512x and higher rates. Light version. |
| ASDM7EC-super 512+fs | Adaptive seventh order one-bit delta-sigma modulatro with extended compensation. Optimized for 512x and higher rates. Super version. |
| AMSDM7 512+fs | Special adaptive seventh order “pseudo-multi-bit” modulator optimized for rates ≥ 20 MHz. |
| AMSDM7EC 512+fs | Special adaptive seventh order “pseudo-multi-bit” modulator with extended compensation optimized for rates ≥ 20 MHz. |
| AHM5EC5L | Experimental fifth order five level hybrid modulator with extended compensation. Optimized for rates ≥ 40 MHz. Note! Limited SNR, best suited for loudspeaker system and/or when digital volume control is not needed. Not recommended when HQPlayer's volume is used as the primary volume control method! |
| AHM7EC5L | Experimental seventh order five level hybrid modulator with extended compensation. Optimized for rates ≥ 40 MHz. Note! Limited SNR, best suited for loudspeaker systems and/or when digital volume control is not needed. Not recommended when HQPlayer's volume is used as the primary volume control method! |
Bit rate
Fixed output bit rate to use, can be used to lock output to a single optimal bit rate. When set to Auto, output rate is highest allowed by DAC capabilities, Rate limit setting and capabilities of the selected filter.
Rate limit
Default preferred output rate and maximum rate for automatic selection.
ALSA backend
Settings for locally connected audio devices.
Device
Output device selection. Currently active device is shown as "Current" since it is in use at the moment. Other devices that are not busy/reserved are shown on the list.
Channel offset
For multichannel devices that expose for example multiple stereo pairs, or have separate channels for different types of output (eg. analog and S/PDIF), offset for the channels to be used is set here. For example if a device has 8 output channels (4 stereo pairs), number channels is set to 2 and output is wanted on channels 3 and 4, offset is set to 2.
DAC bits
Many audio devices claim different PCM word length than the actual D/A conversion has. For example S/PDIF interface could claim to have 32 bits while only 24 bits can be actually transmitted over S/PDIF. And a DAC at the other end of S/PDIF could be R2R design having only 16 bit word length. Since for example S/PDIF is unidirectional, it is not possible to obtain information about the connected DAC. For this reason it is important to set the actual number of bits to be used here. This defines dither / noise-shaping depth for the output. Setting value to 0 means the word length claimed by audio device should be used.
Buffer time
Size of the device driver audio buffer in milliseconds. 100 ms is good starting point for the value and there is rarely need to change this. Setting value to 0 means default buffer size.
DoP
For DSD capable devices that are not supported for native raw DSD transfer, usually support DoP method where DSD data is packed into regular looking PCM samples with special marker bytes the DAC recognizes and knows how to decode the data. Do not enable this setting if you are not sure your DAC supports DoP. If it is enabled for devices that do not support it, quiet hiss can be heard during playback.
48k DSD
Most DACs that support DSD, only support DSD at multiples of 44.1k base sampling rate. However they still likely announce support also for 48k base DSD sampling rates. This setting enables these rates, only set it if you are sure that your DAC actually supports DSD also at 48k base rates.
Network Audio backend
Settings for Network Audio Adapters (NAA).
Device
Output device selection. Currently active device is shown as "Current" since it is in use at the moment. Other devices that are not busy/reserved are shown on the list. List shows all free audio devices found in NAA's found to be available on the network. List items combine name of the NAA and audio device behind it, separated by a colon.
DAC bits
Many audio devices claim different PCM word length than the actual D/A conversion has. For example S/PDIF interface could claim to have 32 bits while only 24 bits can be actually transmitted over S/PDIF. And a DAC at the other end of S/PDIF could be R2R design having only 16 bit word length. Since for example S/PDIF is unidirectional, it is not possible to obtain information about the connected DAC. For this reason it is important to set the actual number of bits to be used here. This defines dither / noise-shaping depth for the output. Setting value to 0 means the word length claimed by audio device should be used.
Buffer time
Size of the device driver audio buffer in milliseconds. Setting value to 0 means default buffer size. Always use default buffer size, unless you have a strong reason to do otherwise.
DoP
For DSD capable devices that are not supported for native raw DSD transfer, usually support DoP method where DSD data is packed into regular looking PCM samples with special marker bytes the DAC recognizes and knows how to decode the data. Do not enable this setting if you are not sure your DAC supports DoP. If it is enabled for devices that do not support it, quiet hiss can be heard during playback.
48k DSD
Most DACs that support DSD, only support DSD at multiples of 44.1k base sampling rate. However they still likely announce support also for 48k base DSD sampling rates. This setting enables these rates, only set it if you are sure that your DAC actually supports DSD also at 48k base rates.
IPv6
Enable/disable IPv6 support. In most cases can be safely left enabled. When IPv6 support is enabled, devices having only IPv4 connectivity can be still found and used. When IPv6 support is enabled, network is used in dual stack mode with support for both IPv4 and IPv6 simultaneously. NAA discovery usually works more reliably when IPv6 support is enabled. IPv6 supports auto-configuration for local networks.