Musiland has introduced a new version of their MONITOR series of USB audio interfaces. The Monitor 03 US will be available in July and incorporates a series of advancements as summarized in the table below.
You can see from the photo the I2S lines (they are on the same pins as before). Like before, easy to tap to feed a higher-end external DAC. Is it worth it to get the new 03 US for modding? Possibly if you also plan to use the DAC and headphone sections are the enhancement are mostly in those areas.
Enhancements in the USB/FPGA sections are small and perhaps not worthwhile to pay 3X the price of a Monitor MINI
Comparison with 02US
|USB Interface||CY7C68013A-56PVXC||CY7C68013A- 56LTXC||Same device, different packaging SSOP vs QFN. Still USB2 speeds since CY is just barely releasing the USB3 version of the device|
||24 Mhz Crystal||24 MHz CTS-CB3 Oscillator
||Phase Jitter (12kHz-20MHz): < 1 ps RMS. Functionally it is the same as before, it provides the reference clock for the CY USB chip which then feeds the FPGA. Using an oscillator will likely result in lower overall jitter.The clock in my device is a 50ppm, 3.3V part
|FPGA||XC3A50A||XC3S200A||The larger FPGA has 4 DCMs that can be dedicated to generate the 44.1 and 48K family of sample rates. Rather than reloading the multipliers every time there is a change in sample frequency family, they can be loaded at all times enabling fast switching of sample frequency [How the clocks are generated]. Thus there is no “fast” and “precision” Sample Rate Control modes in the Musiland control panel|
|DAC||PCM1793||PCM1798||Dynamic Range: 113 dB vs 123 dB
THD+N: 0.001% vs 0.0005%The PCM1798 is capable of supporting 384Khz sample rate even if not specified in the data sheet. Thus 384Khz SR is supported without resampling all the way up to the DAC
Noise: 6 nV/√Hz vs 4.5 nV/ Hz
Distortion: 0.0006% vs 0.002%
Slew Rate: 22 V/µs vs 7.0 V/µs
Bandwidth: 9 MHz vs 16 MHz
Perhaps the weak link of the entire design? MC33079 is not a well known “audio” device (say as compared to the OPA 4134). See section below for Musiland’s point of view
|Head Amp||None||TPA6120||03 has a high performance headphone amp: SNR: 120 dB; THD+N: 0.00014%. This is also used in other “high-end” designs|
|Output||Toslink, Coax||Toslink, MULINK||Mulink is a Musiland proprietary interface for 32 bit transfer and beyond. It is here to connect to their DACs and future technologies|
|USB Interface||USB2||USB3||03 has a USB3 connector, leveraging the higher power of USB3 (900mA vs 500 mA). The interface speed is still USB2 because of the CY chip used.|
|Power||Mains DC-DC supply||USB DC-DC supply||According to Musiland, “new technology” has been developed for DC-DC step-uo, step-down and regulation in order to all the electronics to >90% of their performance from USB power. The first implementation of this technology is the Musiland 03 US|
|Other||-||EEPROM||There is an I2C EEPROM chip on the backside of the board|
|Street Price||$125||$160||Already available at TaoBao for about 200 RMBs more than the Musiland 02US. Also available from Tam Audio|
|More Photos||Soomal||Soomal||Overall, you can see an advancement of manufacturing technology with the use of finer-featured devices|
About the choice of opamp
As shown in the table, every component in this new device is “top notch” and well known in the audio circles, except of the opamp.
The new opamp used for I/V conversion has lower noise, but higher distortion as compared to the previous device and especially compared to popular opamps such as the OPA4134.
In addition, the slew rate is also lower: 7 V/us vs 20V/us. Should this be a “concern”? As explained here, even 7 V/us is an overkill for up to 20KHz audio content:
SLEW RATE: Despite what you may have heard or read you only need 0.2 V/uS slew rate per volt of RMS output to perfectly reproduce any signal you’ll ever find on a CD. That number has been verified by several well respected audio engineers like Douglas Self. And it’s conservative–It assumes a worst case full output at 20 Khz which pretty much never happens anywhere but on a test bench. The E7 has about 1.8 volts maximum output, so it needs 0.36 V/uS for the absolute worst case.
Here is another observation from the same blog:
…Any op amp with a slew rate of 3 V/uS or greater is fast enough for nearly any audio application on the planet. And op amps like the 5532 can easily have bandwidths out to 200+ Khz in most applications which results in negligible phase shift or “delays” in the audio band.
I asked the question of opmap choice at the Musiland forums. This is the answer I got:
Why not use OPA4134 instead of MC33079?
It seems every component is top class for audio except MC33 …
glt 发表于 2011-7-2 00:33
Basically the MC33079 is a better fit to their system as compared to other “more popular” opamps such as the OPA4134. Key in their research is the “small signal” behavior and in particular the offset voltage. Looking at their data sheet, the OPA4134 has an offset voltage of 0.5mV whereas the MC3309 has an offset voltage of 0.15 mV…
Input offset voltage translate into errors in an opamp circuit. According to this entry:
Another practical concern for op-amp performance is voltage offset. That is, effect of having the output voltage something other than zero volts when the two input terminals are shorted together. Remember that operational amplifiers are differential amplifiers above all: they’re supposed to amplify the difference in voltage between the two input connections and nothing more. When that input voltage difference is exactly zero volts, we would (ideally) expect to have exactly zero volts present on the output.
I suppose as a head amp, it is important to care about noise and offset voltage especially because the device is seldom used at full output (as in the case of a DAC alone) and the volume is controlled in the digital domain. In this two parameters, the MC opamp has better specifications than the OPA part. I believe the choice made by Musiland on the MC opamp makes good engineering sense.
Here is a photo of my own Musiland 03:
8/30/11: Becoming more available. You can purchase in Hong Kong for HK$1080 That is about US$ 140
I don’t know if there is a well defined direct correlation, but this setting gave me a pretty stable lock with DPLL set at “LOW” with the Musiland USB -> I2S. I am pretty happy with the performance of this USB interface so far.
Seems ESS designed the DPLL with low enough bandwidth to be a good experimental gauge of jitter for even the big league USB interfaces.
I can probably do a more scientific experiment by wiring the lock LED to an arduino interrupt pin and count the unlocks within a period of hrs…
More on Fiio D3
3.3V or 5V OPERATION?
According to the datasheet, the Cirrus 4344 DAC can operate at 3.3V and at 5V. At 5V operation, you get 2 dB better performance pretty much across the board.
The current operation of the D3 (original model), seems to be operating at 3.3v since the input voltage to the device is 5V.
However, by measuring the voltage on the power pin of the DAC (VA, Pin 9), the DAC is actually operating at 4.8V which is pretty much ideal for best performance.
In addition, the output stage opamp is also operating at the same 4.8V. Therefore, there is no need to do any modding to get better performance.
But a good and easy mod here is the input capacitor. Space is very tight, must use similar sized capacitor. Used a standard class ELNA 1000 uF capacitor, bypassed with a 22nF film cap. The original was 470 uF, 16V.
Detail after removing existing capacitor
After the large capacitor, power goes through a ferrite, then it is bypassed by two small value ceramic capacitors and connects to V+ in the opamp
Detail after replacing capacitor and bypassed with film cap
NEW VERSION: D3K TAISHAN
WM8805 replaced by Cirrus CS8416
As FIIO shared when they first introduced the D3, the WM8805 (perhaps) has been discontinued and they have replaced the chip with the CS8416. This is the new “D3K TAISHAN” model. (Taishan as in “Mount Tai”, a famous place in China where emperors used to worship the heaven and to pray for peace and prosperity)
A reader shared and internal photo (see comment #40):
Comparing with the original version, the layout has been completely redesigned. The DAC and the opamp remains the same.:
Backside of D3K Taishan edition. Date is 11 months after the original V1 version.
Same reader alerted me that there is “in-between” version between the original D3 and the new D3K. We call it D3V2. This version uses the WM8805 part but there are omitted components in (what I believe) is the power section for the opamp. You can check the comments for comments about the performance of this version.
I’ve made a composite image of the D3 Original and the “D3 V2″ which is shown below:
Here is the backside of the D3 V2:
The case is identical with V1. In the V1, the RCA connectors are silver color. The RCA connectors in V2 are gold plated. (V2 is the one to avoid)
Update 2/14/12: More information on the DAC, I2S to External DAC, see end of this post
WM8805-BASED DAC FOR $30
I was intrigued by this DAC based on the industry-leading Wolfson SPDIF receiver WM8805. According to the specifications it is capable of 24bit/192KHz operation and it only costs $30. (Even less nowadays)
Just like any good DIYer, the first thing I did was to take it apart…
The famous Wolfson WM8805 SPDIF receiver. Even though it is capable of muxing up to 8 SPDIF sources, it is configured in hardware mode and therefore it is configured to have a single input into the receiver. The device has a coax and a toslink input and these are selected with a single pole switch. Notice also the use of a seam-sealed crystal instead of the more traditional can (most likely for size).
The WM8805 interfaces a Cirrus Logic 4344 DAC (“344″ indicates fixed I2S configuration). This DAC has pretty decent specs at 105 db SNR and -90 db THD+N, a cut above the DACs found in this price range (in reality, you can’t find any DACs at this price range which is a new low). The CS4344 DAC is also found in the Apple Airport Express 802.11n version.
The analog output of the DAC connects to a TI LMV358 opamp
You can see the I2S lines (the 4 diagonal traces), ready to be tapped…
The power is supplied by an external 5V DC switching supply through a standard mini-USB connector. The 5V line (the uppermost trace from the USB connector) connects to a 6.8 (R37) ohm resistor and to the main PS capacitor (470 uF 16V). This provides a first stage RC filtering to the incoming power.
The 5V incoming is filtered through L5 and further regulated with a simple discrete Zener diode circuit (basically a shunt regulator) to 3.3V which feeds the DVDD (pin 1, digital core suppy) of the WM8805. This is pretty good filtering and regulation. This line also feeds PVDD (Pin 11, PLL supply) filtered through L2, and feeds the Toslink module through L1. Good use of ferrites for noise filtering.
The zener regulated line also feeds the DAC, with a simple cap bypass. The 102 resistor you see in the photo powers the LED to show Power-on condition. One of the advantages of the 4344 DAC is that it requires minimal external components. In the case of power, it only has a single power line.
I2S to External DAC
More on the CS 4344 DAC
The CS 4344 DAC is not a very popular DAC among audiophiles. It is used in the Apple Airport Express (the 10-legged device near the center of the photo), a device not always appreciated by audio enthusiasts.
The New (2012) Airport Express changed to a new DAC:
According to RogueAmoeba:
More pertinent to our customers, the audio digital to analog converter is an all-new 24-bit/192khz Asahi Kasei AKM4430. This chip is similar to the Cirrus Logic CS4344 used in the previous model, but should be a improvement over the Burr-Brown PCM2705 used in the original 802.11g Airport Expresses
And also used in the FubarIV Plus [link].
More on the OPAmp
A user reported changing the opamp to a AD8656 with great results. The AD8656 is also a recommended component of the Gamma-2 DAC and it can be used to drive headphones directly. The OPA2209 is also another pin-compatible rail-to-rail opamp with even better noise specification than the AD part. It is used in TPA’s Trident shunt regulator.
|Input offset (max)
||9 mV||0.25 mV
|Input Voltage Noise Density
An excellent report on the device with high resolution photos (very nice photos), extensive measurements and listening impressions [link].
BetterSound is an alternative to Audionirvana, Pure Music, Amarra…
I use Windows 7 and iTunes. Although I cannot take advantage of this new player, I think this is a great development for the audio community
“At least as good as Audionirvana”
…compared it against Amarra in playlist/cache mode and actually like BetterSound better. It has a more open sound than Amarra and Audirvana. The soundstage is both wider and deeper with better instrument placement. It also seemed to run as good as iTunes itself. Don’t think you’d even realize this was playing in the background. With Amarra, there was always some delay and it couldn’t do gapless with loading whole album into playlist. Audirvana worked great but had to deal with separate player…
New versions will be announced in the thread until the author submits the application to the Apple App Store.
- Exclusive mode
- Do not apply sample rate conversion
- The above two features enables automatic sample rate switching in iTunes, one of the most lacking features of iTunes
- Two 128MB buffers for playback (good for 3 minutes of 32/96K content)
- Focus on performance with highly optimized code (minimized thread locking overhead, manual memory management, others)
- FLAC support (through “Fluke”)
In my opinion, the above features will match all the (bit-perfect) s/w players out there in terms of performance and transparency.
- Arbitrary number of named inputs
- All settings can be customized per input
- All settings are saved in EEPROM and last setting remembered after a power cycle
- “Soft mute” control from the console and from the remote control
- Check the CODE tab
- Can display (and select) two additional settings: IIR filter selection and notch delay.
- Cleaner (less “busy”) and better organized display
- Check the code tab
He played 24bit/352khz material with a SDTrans transport through a Bufflao II DAC with an 80 MHz clock.
“The result was good, I got nearly “clean” DXD playback – there’s still very slight noises, but much less than when using the stock [firmware in the] MC”
- Sharp roll-off filter: Too glitchy, never locks
- Slow roll-off filter, 6bit quantizer: Can play music without unlock, but there’s also annoying white noise.
- Slow roll off mode, 8bit quantizer: Can play music, and less noise
“pseudo differencial mode” does not seem to work properly in my setup, so I used 8bit quantizer in true differential mode.
This seems to be (I am guessing here) an indication that the reduction of “out of band” noise afforded by the higher bit length of the quantizer has an audible effect at least with ultra high sample rate material.
According to Dustin Forman of ESS Tech, on one of his postings at diyaudio:
One way to add DACs together is to simply duplicate the input to many DACs and sum them up at the output. This buys 3dB DNR improvement every time you double the amount of DACs. Basically uncorrelated noise adds RSS and signals add normally. Another way would be to use a larger bit QUANTIZER and route the signals from the QUANTIZER to 2 DACs, but the noise shaper now has an extra bit in it.
This chip does both. You can 1- simple duplicate the data input header and then add up the outputs in an analog circuit on the board, or 2- you can program the chip to use a larger QUANTIZER. Doing this prevents the need to send the same data to all the inputs since now a certain DAC Channel is routed into 2 DAC outputs (one channel into two channels).
The DAC is normally a 6-bit QUANTIZER, with the DACx being the summation of the 6 bits, and DACxB being the summation of the inverse of the SAME 6 bits. This is the best all round performance mode, and this is why the datasheet says register 15 needs to be set to 8′b00000000.
Setting this register to 8′b01010101, which by the way was the mode I thought would work best base on my prototype design, (and that is why it is the default configuration) the DAC becomes a 7-bit QUANTIZER reducing out of band noise. I simple divide up the 7-bit number coming from the QUANTIZER into 2 6-bit numbers and invert 1 of them. Then I send off these new 2 6-bit numbers which the difference is mathematically identical to the original 7-bit number from the QUANTIZER and ship them off to the analog section. This also results in 8 channels at the input being routed to 8 channels at the output.
Now, let’s go further (we are only 1/2 down this road). if you set the register 15 to be 8′b10101010 then you get a DAC with an 8-bit QUANTIZER, out of band noise decreases more and so on. Now I shut off 1/2 the internal logic since it is not required, only inputs 1,2,5,6 are now needed since an 8-bit output can be spliced into four 6-bits numbers. Channel 1 is merged with channel 3, 2 is merged with 4, 5 is merged with 7, 6 is merged with 8. This arrangement is to keep the merged channels analog sections as close as possible for device matching inside the chip.
This gobbles up 2 analog sections [You still have 2 analog sections per DAC-channel active, but ½ of them take the input from another DAC-channel as indicated above] per input now. This is why 1/2 the digital section is shut off. So this can make you a 4 channel DAC while putting data into only the first 4 channels. [4 Channel -> 4 DAC-Channels -> 8 DACs (8-bit quantizer) -> 16 analog sections]
Ok, let’s go further, how about a 9-bit QUANTIZER? sure why not. Setting register 15 to 8′b11111111, I shut off 6 of the channels internally and only channels 1 and 2 inputs are routed to the analog sections. Well it is probably obvious by now why, but here it is again: 1 9-bit number can be broken into 8 6-bit numbers. Now route the 8 6-bit numbers to the analog sections (remember that each section is DAC and DACB so there are 2 analog sections per DAC or 16 total in the chip) so now with each input taking 8 analog sections, we have 2 channels.
Put a new version of the code (v07d) in the CODE section
- Tracks DPLL setting separately for I2S and SPDIF allowing fine tuning of the DPLL setting depending on the input format
- The DPLL settings still default to “best” for I2S and “lowest” for SPDIF at power-on. These have shown to be the best stable settings especially when the DAC is cold
Added Selection of quantizer and other improvements:
- 6-bit True differential
- 7-bit Pseudo differential
- 7-bit True differential
- 8-bit Pseudo differential
- 8-bit True differential
- 9-bit Pseudo differential
What does changing this setting do?
- Decreases out of band noise as explained by the chip designer at diyaudio [link]. This allows the design of the analog section (the analog output stage such as Legato) to “run with a less aggressive filter [which] has some nice benefits (better slew rate etc) to the final audio” [link]
- Shuts off certain digital sections of the DAC. In 8-bit mode, for example, the digital sections of 4 DACs (half of the eight DACs) are merged and the unused half is shut off. I will speculate here that this reduces internal noise and interference.
Where is the code?
Visit the CODE tab
I can’t hear differences between the different settings, but some of you may hear differences. Please post your experiences…
NOTE: update your Arduino s/w to version 0022 (or later)