True 32 bits of SPDIF 768khz :
ComTrue Inc. - Products
.SPDIF 768k/32
.PCM 768k/32
.DSD 8x
.Dop 4x
.ASRC/SRC
ComTrue Inc. - Products
.SPDIF 768k/32
.PCM 768k/32
.DSD 8x
.Dop 4x
.ASRC/SRC
Do you know if it supports +0dBFS signals? (intersample peaks over 0dBFS)
http://www.diyaudio.com/forums/digi...amp-crossover-dac-project-40.html#post4476464
I see you have the ASRC before the volume control.
Maybe thre will be inexpensive digital volum and samplerate controls now, for us that is not affraid of digital attunation but want to do it outside the PC.
Regards Torgeir
http://www.diyaudio.com/forums/digi...amp-crossover-dac-project-40.html#post4476464
I see you have the ASRC before the volume control.
Maybe thre will be inexpensive digital volum and samplerate controls now, for us that is not affraid of digital attunation but want to do it outside the PC.
Regards Torgeir
Do you know if it supports +0dBFS signals? (intersample peaks over 0dBFS)
http://www.diyaudio.com/forums/digi...amp-crossover-dac-project-40.html#post4476464
I see you have the ASRC before the volume control.
Maybe thre will be inexpensive digital volum and samplerate controls now, for us that is not affraid of digital attunation but want to do it outside the PC.
Regards Torgeir
input signal range = 0x7fff_ffff ~ 0x8000_0000 (sign)
ASRC = 3-stage FIR filter with 64-bit resolution, 0.5 LSB distortion
volume control = +18dB ~ -110dB step 1/32 dB
output signal range = 0x7fff_ffff ~ 0x8000_0000 (sign)
In this case input=sign 32-bit, FIR=sign 64-bit, volume=sign 32-bit,
distortion=0.5 LSB,
and we can get 31.5 bits accuracy.
: I think Ur question is interpolation filter & re-sample issue.
as I know, answer is yes. This chip, CT7302PL, can maintain +0dBFS.
input signal range = 0x7fff_ffff ~ 0x8000_0000 (sign)
output signal range = 0x7fff_ffff ~ 0x8000_0000 (sign)
: I think Ur question is interpolation filter & re-sample issue.
as I know, answer is yes. This chip, CT7302PL, can maintain +0dBFS.
I dont understand how it can handle intersample overload without having higher output signal range than input signal range. My bet is 4 bit resolution and noise at -20dBfs
As stated the problem is that a signal with intersample overloads should be attunated the same amount as the intersample overload max.
Normaly -6dB would do the trick.
I guess two chips in series would solve it:
The first has no SRC been done only -6dB attunation -> Transfer samples at 32 bit -> The second has SCR or ASRC and then digital volum control.
After digital volum control there should be no need for signals at more then -6dBFS with a high quality DAC and propper analog gain before the amps.
If it is a digital amp fed intersample overloaded signals it will clipp them.
Regards Torgeir
Normaly -6dB would do the trick.
I guess two chips in series would solve it:
The first has no SRC been done only -6dB attunation -> Transfer samples at 32 bit -> The second has SCR or ASRC and then digital volum control.
After digital volum control there should be no need for signals at more then -6dBFS with a high quality DAC and propper analog gain before the amps.
If it is a digital amp fed intersample overloaded signals it will clipp them.
Regards Torgeir
I dont understand how it can handle intersample overload without having higher output signal range than input signal range. My bet is 4 bit resolution and noise at -20dBfs
for example :
sample rate = 48khz, single tone
if : input signal = 12khz sin tone, 2pi/4 offset
=>0dB max = 0x7fff_ffff
if : input signal = 12khz sin tone, 2pi/8 offset
=>0dB max = 0x7fff_ffff * sin(2pi/8)
: u may think about frequency domain, not time domain.
not all 0dB max value = numeric max value
I am thinking about the signal enclosed in the posts:
http://www.diyaudio.com/forums/digi...amp-crossover-dac-project-40.html#post4477859
or
http://www.diyaudio.com/forums/digi...re-detect-0dbfs-waveforms-overload-times.html
In your example you have to resample upwards. Then calculate again.
http://www.diyaudio.com/forums/digi...amp-crossover-dac-project-40.html#post4477859
or
http://www.diyaudio.com/forums/digi...re-detect-0dbfs-waveforms-overload-times.html
In your example you have to resample upwards. Then calculate again.
BTW, I really like your product if it lives up to the spec!
And to the non believers in digital volum control bacause of noise
Point 1:
Vn=√(4*Kb*T*R* Δf) (V)
Thermal noise of resistor - Calculator - Audio PerfectionAudio Perfection
Dynamic range re 3V RMS = 144.4 dB for 100ohm
Dynamic range re 3V RMS = 134.4 dB for 1000ohm
Dynamic range re 3V RMS = 124.4 dB for 10k
Dynamic range re 3V RMS = 117.6 dB for 47k
Point 2:
DAC from PCM1798 | Audio DAC | Audio Converters | Online datasheet
Distortion from -10 to 0 dBFS is almost the same relativ to signal.
So source impedance or analog feedback resistors in buffers and analog volum control must be pretty low to gain any advantage from analog volume control.
A firm like RME has abandoned it.
And to the non believers in digital volum control bacause of noise
Point 1:
Vn=√(4*Kb*T*R* Δf) (V)
Thermal noise of resistor - Calculator - Audio PerfectionAudio Perfection
Dynamic range re 3V RMS = 144.4 dB for 100ohm
Dynamic range re 3V RMS = 134.4 dB for 1000ohm
Dynamic range re 3V RMS = 124.4 dB for 10k
Dynamic range re 3V RMS = 117.6 dB for 47k
Point 2:
DAC from PCM1798 | Audio DAC | Audio Converters | Online datasheet
Distortion from -10 to 0 dBFS is almost the same relativ to signal.

So source impedance or analog feedback resistors in buffers and analog volum control must be pretty low to gain any advantage from analog volume control.
A firm like RME has abandoned it.
Last edited:
(I use analog volum control only because I have not found a reliable digital one yet. I mean one that don't say pop/bang or ssssssss at high volume during fault conditions)
I am thinking about the signal enclosed in the posts:
http://www.diyaudio.com/forums/digi...amp-crossover-dac-project-40.html#post4477859
or
http://www.diyaudio.com/forums/digi...re-detect-0dbfs-waveforms-overload-times.html
In your example you have to resample upwards. Then calculate again.
yes.
BTW, I really like your product if it lives up to the spec!
And to the non believers in digital volum control bacause of noise
Point 1:
Vn=√(4*Kb*T*R* Δf) (V)
Thermal noise of resistor - Calculator - Audio PerfectionAudio Perfection
Dynamic range re 3V RMS = 144.4 dB for 100ohm
Dynamic range re 3V RMS = 134.4 dB for 1000ohm
Dynamic range re 3V RMS = 124.4 dB for 10k
Dynamic range re 3V RMS = 117.6 dB for 47k
Point 2:
DAC from PCM1798 | Audio DAC | Audio Converters | Online datasheet
Distortion from -10 to 0 dBFS is almost the same relativ to signal.
![]()
So source impedance or analog feedback resistors in buffers and analog volum control must be pretty low to gain any advantage from analog volume control.
A firm like RME has abandoned it.
yes,
so we always implement R with SC(switch-cap) in VLSI design, not poly-resistor.
yes.
Freq = 12000
SF = 48000
Phase = 45 or PI/4
Max samplevalue without oversampling = +/-1, 0.0 dBFS
=> Max ampl with oversampling = 1.41, +3.0 dBFS
yes,
so we always implement R with SC(switch-cap) in VLSI design, not poly-resistor.
Guess you do not only have all digital products in your product pipeline;-)
Regards Torgeir
True 32 bits of SPDIF 768khz :
.SPDIF 768k/32
.PCM 768k/32
.DSD 8x
.Dop 4x
.ASRC/SRC
Please provide the requested information about the 384/768kHz interface see also http://www.diyaudio.com/forums/digital-line-level/263868-true-32-bits-spdif-interface-new-ic-specification-5.html#post4539226
Hp
Please provide the requested information about the 384/768kHz interface see also http://www.diyaudio.com/forums/digital-line-level/263868-true-32-bits-spdif-interface-new-ic-specification-5.html#post4539226
Hp
ComTrue Inc. - Contact
ComTrue Inc. - Downloads
as i know,
coaxial link : spdif up to 768kHz/32-bit (125Mhz bit rate)
some optical link : spdif up to 384kHz/24-bit (50Mhz bit rate)
Freq = 12000
SF = 48000
Phase = 45 or PI/4
Max samplevalue without oversampling = +/-1, 0.0 dBFS
=> Max ampl with oversampling = 1.41, +3.0 dBFS
yes.
so, source file normalize is very important to make sure frequency domain under "0dB " in signal band (for example,20~20kHz @44.1kHz).
as i know,
coaxial link : spdif up to 768kHz/32-bit (125Mhz bit rate)
some optical link : spdif up to 384kHz/24-bit (50Mhz bit rate)
Fine.. do you have a evaluation board or to show us how you test those speeds. 😀
Hp
Fine.. do you have a evaluation board or to show us how you test those speeds. 😀
Hp
if u need product support.
ComTrue Inc. - Contact
S/PDIF :
source0 = (CD player with spdif output) / (AP2722 32k~192k 16~24bit)
DUT0 = EV-board (spdif in from source 0) (SRC 32k~768k 24/32 bit) (spdif RCA/optical out to DUT1)
DUT1 = EV-board (spdif in from DUT0) (SRC 384k 24/32 bit) (TI pcm5102 DAC out)
speaker = from DUT1 DAC out
AP2722 : from DUT1 spdif out (32k~192k 24bit)
I2S/DSD/Dop :
same as S/PDIF test. just change DUT0/1 I/O port to I2S/DSD or Dop(spdif/i2s)
Last edited:
audio next : digital speaker (spdif in / direct spdif out)
optical : 384k/24 50M bit-rate. distance: i think more than 100m .
(jsr2124, jst2124 : solteamopto)
coaxial : 768k/32 125M bit-rate. distance : we only try 50m, but i think, may
more than 100m .
(RAC connector, 5c2v coaxial cable, 75ohm)
optical : 384k/24 50M bit-rate. distance: i think more than 100m .
(jsr2124, jst2124 : solteamopto)
coaxial : 768k/32 125M bit-rate. distance : we only try 50m, but i think, may
more than 100m .
(RAC connector, 5c2v coaxial cable, 75ohm)
Last edited:
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