It is a great time to be an audio diy’er. There is currently great availability of quality boards aiming at providing the greatest fidelity with incredible VALUE.
Here are some side by side photos of the two USB-I2S interfaces I own.
Worthy of mention is the upcoming next generation Wave IO board. Mr. Lorien posted a sneak peek at his next generation board [link]
From the look of the layout, this board has electrical isolation of the outputs and flip-flop reclocking after the isolator.
Head over to Soomal for a teardown of Korg’s DSD recorder [link]
The new Teac UD-501 DSD capable DAC has received good reviews everywhere. Was curious about its USB interface. Here is a photo taken from a French site, qobuz.com [link]
There is a TMS320 chip and an unidentified chip. It is possible the unidentified chip is the USB receiver or the TMS320 is doing receiver function and the unidentified chip is a microprocessor to control the TMS320 chip.
The TMS320 is a DSP chip [link]. It is a “C67″ series, an up to 1GFLOP 32-bit floating point DSP (the XMOS are rated at 500-1500 MIPS, likely equivalent performance). This family of DSP products was introduced in 1983 and new models have been introduced along the way.
TMS320C6748, a low-power dual-core applications processor based on a fixed-point C64x+™ instruction set and the floating-point C67x+™ instruction set. It provides significantly lower power than other members of the TMS320C6000™ platform of DSPs and provides both floating-point precision and fixed-point performance in the same device. With a wide variety of standard interfaces for connectivity and storage, the C6748 development kit enables developers to easily bring audio, video and other signals onto the board.
… Included in the C6748 development kit is all the hardware and software needed for two demonstrations, a fingerprint-recognition demo and a face-detection demo.
Here is the block diagram for the eval kit:
Looks like a very capable and general purpose processor for not just audio but a lot of other things not even related to audio. You can take a look at the WIKI for all the available libraries for the DSP [link]. The DSP can even implement audio decoding and filtering.
In addition, the USB-2 interface is provided by the FT232 chip (and thus the unidentified chip is likely not the USB interface chip; however, there are no FT232 chips in 48-pin package).
Further, the DAC has an upsampling feature and is provided by the Cirrus CS8422 chip on the main taking the I2S output from the USB board.
Thus the USB board, I believe, it is just moving the bits to the DACs. It looks kind of overkill; perhaps in some future Teac will add other capabilities such such as PCM to DSD conversion and different upsampling algorithm on the DSP chip. It is also likely that Teac is just reusing hardware from their TASCAM proline. A device such as the TASCAM US-366 is a “USB interface with DSP Mixer”
This design from TEAC gives credibility to what Musiland is doing in their upcoming “SuperDSP” chip. The good thing about the Musiland product is that the DSP is dedicated to audio, and the chip will have native support for USB 2 and USB 3 interfaces.
Appreciate Mr. Daussin, the reviewer of the TEAC DAC, providing additional information in the comments section. The unidentified chip is the TPS65070. Readers would recognize “TPS” being power chips. The TPS65070 is a single-chip with multiple voltage outputs. Here it provides the different voltages required by the DSP chip. It is a convenient, integrated solution to provide the power to the main chip instead of using separate regulators.
The TMS320 DSP chip has implemented USB 2.0 capability in s/w and there is no need for a FT USB receiver chip (In the block diagram the USB 2.0 receiving capability is provided by a separate chip because (I think) the DSP chip is already burdened with many different functions).
Here is an English version of Musiland’s recent press regarding the development of a new processor for Audio:
A famous philosopher once said: “Today’s stillness is for tomorrow’s outburst”.
5 years ago, the release of Musiland’s first FPGA-based product, the LILO V ENJOY USB sound card, marked the beginning of Musiland Audio Labs involvement and mastering of chip-level programming technology.
For the last 5 years, Musiland Audio Labs has continue to invest in chip-level programmable solutions by gradually introducing FPGA-based solutions to its entire product line and by the using larger size FPGAs to implement increasingly more complex capabilities.
Four years ago, Musiland Audio Labs recognized that the advances in embedded 32 bit microprocessors, led by companies such as ARM and MIPs, far surpassed the advancements in desktop processors. At that time, Musiland made the decision to use FPGAs (as a bridge) to develop its own general purpose 32-bit processor.
Three years ago, the MD11 was born, followed by the HP11 and MD30. Musiland Audio Labs started to use early versions of its general purpose processor (implemented in FPGA) to handle other functions such as user interface, LCD control and storage. Adding a good graphical user interface to HIFI equipment was just a small step but demonstrated the continued investment towards a general purpose processor technology
Of course the goal to develop a 32-bit general purpose processor is not to just deal with the user interface and other minor tasks. Our vision is to develop a complex, large-scale processor to do more complex audio processing functions such as decoding FLAC. This requires a plan with a long time horizon to develop a processor that not only meets current audio processing needs, but also keeps up with the advancements of computer technology and meets future needs.
On deciding on the architecture for the processor, Musiland engineers had different views on what to do. The technology camp felt that they could develop the processor from scratch (call it “M-CPU”) and thus not be subject to any external constrains or patent protection; but more seasoned engineers knew that a development from scratch was a long and sky-high expensive proposition. After exploring different proposals and after much discussion among the different teams, a rational, consistent and scientific decision was reached: Use ARM+DSP dual-core architecture.
Industry leaders had same strategy. At the same time Musiland Audio Labs decided to create a 32-bit processor based on ARM+DSP dual core, we learned that Creative Technology set up Zii Labs for the research and development of multimedia processors. This was a great encouragement for our Musiland team and allowed us to understand the competitive environment with more clarity and also to strategize on how and where to differentiate our products. We decided to abandon complex video functions and to focus instead on the field of audio.
Later on we lamented the divesture and selling of Zii Labs but realized that our decision to focus on audio was the right decision. We at Musiland Audio Labs will always respect the people at Creative, and would like to take this opportunity to appreciate Creative’s contribution to the industry and in making multimedia and audio applications and products so pervasive. We must always remember the name Zii Labs for their truly innovative processors.
For the past two years, Musiland Audio Labs engineers worked night and day to integrate the ARM core and audio DSP functions and developed an ARM-based 32-bit processor with 32-bit floating point DSP with an internal unique data bus, the “MP-Bus”. We will call this new processor “SuperDSP” and will produce a family of products: SuperDSP100, SuperDSP200, etc.
A family of devices will be produced to meet different applications. The different versions will differ on the size of on-board and external memory, the type of packaging (QFP and QFN). The chips will support USB 2.0 high-speed and optionally USB 3.0, external mass storage support (SD, MMC), external bus support (PCI) and other communication options such as SPI, I2C UART, etc.
It is worth mentioning that the 32Bit floating point audio DSP unit can handle up to 64Bit/768kHz sampling rate of the audio data, support multichannel or DSD decoding.
According to our product roadmap, the first SuperDSP processors will be implemented for the personal multimedia sound card/integrated player market and will be released towards the middle of this year and will provide unprecedented audio capabilities and sound experience. Soon after, we will focus on the HIFI and audiophile products. So stay tuned!
If you have any questions, requirements or expectations about this new product, please use our forum (bbs.musiland.cn- look for SuperDSP) to post your queries. Meeting your requirements is our responsibility; be sure to let us know!
The SuperDSP interface, decoding algorithms and Library API will be open to third party developers. We welcome industry colleagues to discuss OEM/ODM arrangements with us. Musiland Audio Labs looks forward to this cooperation and is prepared to help you innovate your products
Follow the discussion of this board and XMOS technology in general in the diyaudio thread: [link].
The new board
The new chip from XMOS
Cannot tell from the chip markings what part is it. It is “missing” the part number. But it is definitely a “U” part as there is no separate USB controller chip. Here is the datasheet of one version of the part: [link]
The previous generation XMOS devices required a separate USB interface chip (foto from here: [link]):
The audio clocks. According to the datasheet, pin 1 is the enable pin and pin 3 is the output pin. Looks like the enable pin is controlled by the XMOS chip, enabling/disabling the clocks for the corresponding sample rate.
The local ultra low noise LDO (excuse the left over cotton fibers from my cleaning ). You can bypass the USB power by removing FB1 and provide external 5V.
The 2 LC filters for filtering the output from the built in DC-DC converters. Here is place to add larger caps and increase the filtering.
Solid ground plane. CN1 is for external powering (instead of using USB power).
If you read the datasheet and compare the topology of this board you will find that the manufacturer has followed the build and layout recommendations from XMOS. In fact it is so simple, that not following the manufacturer’s recommendations is kind of hard
The new XMOS chip has several advantages over the old device
- Built-in USB receiver
- Requires much less external components and thus it has been optimized to be implemented on a 2-layer board
- Built in voltage regulators. The internally generated regulated power is further filtered by external LC filters – Here is an area to mod: add capacitance to increase the filtering
- The clocks can be positioned much closer to the device than in the previous design
Apparently 384KHz support was added by diyinhk according to this lively discussion at diyaudio [link]
FEATURES OF THE MODULE (From the manufacturer)
- 6.5uVrms Ultralow noise LDO. -The TI LP5900 [link]
- Solid ground plane (a must for high speed digital circuit)
- No Via in active circuit (via inductance always create jitter problem)
- FOX Xpresso ultra low ppm oscillators and Murata capacitors sourced from Digikey USA. -The two Xpresso 22.579 2MHz and 24.576 MHz clocks. Not the ultimate in low jitter, but pretty low jitter. See the graph below
- Gold plated USB connector
- Compact size 50mm x 30mm. -That is even smaller than the Amanero which is 30×70 mm approx.
- USB powered but can be externally powered by removing FB1 and connect 5V to CN1 (warning: over-voltage or reverse-voltage can damage the XMOS chip immediately. Any modification will void item warranty)
- Can optionally install series resistor to I2S lines (by cutting the traces)
Comparing the Xpresso clocks with Crystek.
I first checked it by plugging it to a Macintosh computer. Native MacOS supports up to 352K and 384K sample rate.
On the PC, it requires device drivers.
Version 1.63.0 is the latest driver from Thesycon.
No indication of 352K/384K capability for WASAPI shared mode. This may in fact a limitation of Windows mixer as it expects to resample every audio stream to the selected sample rate. However, in WASAPI exclusive mode, this part (the mixer) is totally bypassed and the output sample rate from the player application is passed directly to the hardware.
The driver also supports ASIO
CONNECTING TO DAC
Connected the board to my old Buffalo II DAC (80 MHz clock) and tested all sample rates from 44.1KHz all the way to 384KHz. With the 80MHz version of the Buffalo II DAC, 352.8KHz and 384KHz work fine with oversampling turned off. All sample rates work as advertised. Modern designs are pretty robust, especially this version of the XMOS chip which requires only a few external components.
The socket in the XMOS device matches the socket pin arrangement for the latest version of the $99 ES9018 board and a ribbon cable can be used. For the Buffalo simple cat-5 twisted pairs can be used.
Here is an implementation from a reader with diyinhk $99 DAC board:
The XMOS is hiding…
Straight from the eBay page [link]
I have to congratulate this Mr. diyinhk. He can non only produce designs with such speed, but can somehow price them so amazingly low. This new module is priced at $59.95 -Can’t resist not getting it!
- Latest XMOS chip that uses 48MHz oscillator (rather than the 13Mhz used by the older XMOS chip for the USB interface) – I believe this is the XS1-U chip with built-in USB interface [link]
- 6.5uVrms Ultralow noise LDO. -The TI LP5900 [link]. I believe (the 6-legged chip next to the USB connector).
- Solid ground plane (a must for high speed digital circuit)
- No Via in active circuit (via inductance always create jitter problem)
- FOX ultra low ppm oscillators and Murata capacitors sourced from Digikey USA. -The two Xpresso 22.579 2MHz and 24.576 MHz clocks. They are of different size which reminds me of user complaining when some of the Hiface interfaces were built with the smaller size oscillators. But the industry is moving away from 5×7 mm to 3×5 mm and there is no way back.
- Gold plated USB connector (Molex, FCI, or other -depends on stock)
- Compact size 50mm x 30mm. -That is even smaller than the Amanero which is 30×70 mm approx.
- Module is USB powered but the diyer expert can remove FB1 and use an external 5V PS to CN1 (warning: over-voltage or reverse-voltage can damage the XMOS chip immediately. Any modification will void item warranty)
- Can optionally install series resistor to I2S lines (by cutting the traces)
WINDOWS DRIVERS PROVIDED (NO DRIVER REQUIRED FOR MACS)
Looks like the first USB interface using the latest generation XMOS chip (XCORE USB) with built-in USB.
Just downloaded the latest version of iTunes. Noticed a new playback panel:
Seems you can up-sample your content within iTunes using Apple’s SRC. I did a quick comparison vs 44.1K/16bit and preferred the up-sampled playback… I realize that you can’t add data back to the original file, but playing at a higher sample rate will change the digital filtering applied in the DAC.
To allow direct playback of the iTunes up-sampled stream, you need to match the sample rate in the sound control panel. Otherwise, Windows will re-sample the stream to match the setting. This is because iTunes only supports WASAPI SHARED mode.
Having the Hifiduino controller show the incoming sample rate, helps tremendously in the set up and configuration of different audio players and their options.
Here is the Playback panel of the previous version
Apple has yet to document the new playback panel in the latest iTunes 11.0.2. If you search their website or the help files, there is no mention of this new option. Maybe this is work in progress and there is more to come in newer versions of iTunes. As it is, it works straightforwardly by determining the audio output sample bit rate and bit depth during playback.
Since iTunes 10.5, the Quicktime component is not included in the installation [link]. This means that this new playback bit-rate feature is NOT quite a replacement for the bit-rate options in Quicktime (since Quicktime has not been required for a while).
Below is the Quicktime control panel showing the Sound Out bit rate control which is only available with Direct Sound. If choosing WASAPI, no bit rate control is available.
Which begs the question: without the Quicktime control, what default mode was iTunes using?
Having never supported WASAPI exclusive mode, iTunes will re-sample every track to the sample rate set by the player. Audio purists will immediately see this a not bit perfect. However, upsampling everything may not be a bad thing. Benchmark Media has been encouraging using iTunes upsampling for some time now [link]:
Sample rate can be set to match the sample rate of the media or to the highest that the audio interface is capable, since the upsampling in iTunes is harmless… The user should not be discouraged from setting the sample-rate to 96 kHz as a permanent setting, even when the audio is less then 96 kHz.
Further, the Sabre32-based DAC, seems to have a sweet spot with higher input sample rate. Thus the higher sampling rate seems to be specially beneficial to Sabre32-based DACs.
MORE LISTENING TESTS
I spent some more time comparing 11.0.2 iTunes playback at “bit-perfect”, in this case 44.1K material vs. upsampled to 192K/24
Even though I am not one that has the capability to detect minute differences (for example by changing a passive component such as a capacitor or a power supply), this time I was able to readily hear differences between the two playback sample rates. In all cases, I preferred the upsampled playback. This says that the differences are not minute.
The 192K/24 playback provided the following improvements:
- A more 3D soundstage. Some would say more “holographic”
- More detailed resolving of the sound. The same sounds, but “finer”, more “delicate”, or some may say “super high resolution”.
- The harmonics gave a “greater presence” of the music. Perhaps greater dynamics.
EFFECT OF FILTERS?
I am sure there is no magic with the Apple upsampling algorithm, and there is no way to add information that in not there to start with. The only other explanation for the improvement is the use of different filters in the Sabre32 DAC.
This is an update to this post on using foobar to play both PCM and DSD files [link]
Playing PCM and DSD files with foobar:
After installing foobar2000,
From the foobar2000 website, download:
- ASIO support 2.1.2 or newer
Get to the “Super Audio CD Decoder” repository [link]
- Download the latest file from the foo_input_sacd folder
- Download the latest file from the foo_dsd_asio folder
When you extract the zip file you will find:
- ASIOProxyInstall-x.x.x (application). This installs foo_dsd_asio
This plug in shows up as “Super Audio CD Decoder” and allows you to playback the following formats:
- Playback of Super Audio CD ISO images, DSDIFF files and DSF files
- Converts DSD to PCM (if your DAC cannot play native DSD)
To install or update foo_input_sacd, just drag it to the components window in foobar and click “apply”
The control panel looks like this: you can select DSD output or PCM output (when converting DSD to PCM)
More on the different options: [link]
I tested the conversion of DSD to PCM, using both DSD64 and DSD128 files. The results were as expected:
||Output Sample Rate for DSD64 Input
||Output Sample Rate for DSD128 Input
foo_dsd_asio is kind of a “meta-asio” driver. It is used by the sacd plug-in above as the output device and in turn foo_dsd_asio outputs to an ASIO driver for an actual device. Apparently the sacd plugin does not output directly to an ASIO driver, it must go through the meta-asio driver. (Actually, ASIO itself is a meta-device driver because it talks to the actual device driver)
For one thing, foo_dsd_asio handles the “DSD playback method”: for example, “DoP Marker 0×05/0xFA”. “DoP” means DSD over PCM. Marker 0×05/0xFA means use the marker (for DSD) as specified in the proposed “USB Link for DSD Audio via PCM Frames” open standard [link][link]. This “marker” method is predominantly driven by the MacOS since its built-in USB2 audio driver only supports PCM. On the PC side, there is no native support for USB2 audio so people use ASIO and ASIO can support both PCM and DSD streams. The Amanero board handles “ASIO native”, but other boards may require a marker on the DSD stream.
Thus, foo_dsd_asio has the following functionality:
- Handles DSD playback method
- Converts PCM to DSD (optional)
- Outputs to DSD-capable ASIO driver
To install or update foo_dsd_asio, run the ASIOProxyInstall-0.6.x.exe program.
I tested the PCM to DSD conversion. The latest foo_dsd_asio plug-in handles all sample rates as shown in the following results:
|PCM Sample Rate
AMANERO FIRMWARE/DRIVER UPDATE
Recall that previously ASIO4ALL was used because the Amanero lacked native ASIO support. Amanero updated the firmware and released ASIO drivers.
For instructions on enabling native ASIO playback, you can follow this excellent post:
After updating the Amanero device with ASIO support you do the following:
Fist set the output to be foo_dsd_asio (in order to support DSD output, and also PCM since this is the output device)
And then in the foo_dsd_asio configuration screen, select the Amanero board.
TIPS FOR UPDATING FIRMWARE
1- Plug board into USB port
2- Short the pads as shown for at least 1 second (I used a paper clip)
3- Unplug the board
Download and Install Atmel device driver
1- Download and unzip the update tools from the Amanero web site [link]. You will get several files: a device driver for the ATMEL chip, a configuration application for updating the firmware and some other files.
2-Plug the board into a USB port. At this point the device is completely unidentified. You may get a prompt to install a driver or you may not. Following is the manual installation of the ATMEL device driver:
3- Open the Device Manger under the System control panel. The device shows up as “unknown device” as shown below
4- Right click on the unknown device and select Update Driver Software, indicating the location of the driver which is in the folder you downloaded. Your Amanero board is now a “AT91 USB to Serial Converter”
Your are ready to update firmware
1- Ensure that you have erased the firmware as shown above, and ensure that the device is identified as “AT91 USB” as a port as shown below:
Got a hold of DIYINHK’s C-Media based USB-I2S interface a while back, and it is just now I got around writing something about it.
This is based on the newest USB2 audio chip on the block, The C-Media chip is the CM6631A. This version in turn is the newer, the “A” version of the CM6631 chip which has been implemented in several recent products. The CM6631A differs from the more common (and older) CM6631 in that it supports 176.4KHz sample rate. All other features are the same as the CM6631.
According to the product info page [link]:
USB 2.0 Asynchronous operation (every one does this nowadays)
Up to 192KHz / 32bit
Very capable set of input/output (the implementation reviewed here only implements USB input and I2S output)
- 2 pairs I2S or Left-Justified serial audio output interface
- 2 pairs I2S or Left-Justified serial audio input interface
- Built-in 192K/176.4K/96K/88.2K/48K/44.1KHz and 16/24-bit SPDIF transmitter
- Integrated 192K/176.4K/96K/88.2K/48K/44.1K and 16/24-bit SPDIF receiver
- Supports SPDIF IN-to-OUT loop-back path for signal transforming between TOSLINK and RCA connections
I was able to find an older version of the datasheet for the CM6631 which is pin compatible with the DM6631A [CM6631_Datasheet_v0.8]
It can be seen from the specifications that a more capable system can be developed. For example, a 4-channel I2S output driving two DACs. (allowing performing digital crossover functions in the PC)
DIYINHK USB Interface: CM6631A (with the “A”)
Bottom. Uses the CM6631A
The vendor provides a device driver [link]
Other sources: the latest version of the driver is a unified version for the CM6631 and CM6631A devices, thus available from other manufacturers using either the 6631 or 6631A part (likely they all offer the same versions. some manufacturer may be faster than others in offering newer versions):
I tried the Emotiva provided driver and it works. Here is a screen shot:
There are two versions of the firmware as discussed here [link] and here [link]. However, these versions seem to be related to the SPDIF features of the chip. For this board, which only supports I2S, I don’t think these versions of the firmware matters. Note: according the tdtsai [diyaudio], the developer for the driver s/w, the firmware for the CM6631 is not compatible for the CM6631A.
- Firmware 0101 PID 0×0319 can output to SPDIF via CMedia ASIO, but it will never passthrough DTS and AC3. I2S and SPDIF-Out work simultaneously
- Firmware 0108 PID 0×0314, using current drivers, will not output to SPDIF via ASIO, but it can passthrough DTS and AC3 correctly. I2S and SPDIF-Out are reported as different devices (so one output at a time depending on which one you choose)
Two ultra low noise regulators: the TI LP5900 [link]
Uses high frequency 45.1584 MHz and 49.152 MHz oscillators to derive the audio frequencies (the manufacturer indicates that these are sourced from Digikey, thus ensuring their quality). This is unique as most devices use half that frequency or lower.
No markings? According to the datasheet, this seems normal. The product number is in the box or reel.
The oscillators output are available to be used as master clock for the DAC. The higher speed oscillator are in the “sweet spot” frequencies for ESS Sabre DACs for synchronous operation.
Since the board does not switch the clock lines, you can only select one of the clock lines to feed the ESS DAC. In this manner, with some of the sample frequencies the DAC will operate in synchronous mode and with other sample frequencies, in the normal asynchronous mode.
For example, if you chose to use the X45 line (45.1584 MHz) as the clock for the ESS DAC, then when playing 44.1K, 88.2K and 176.4K the DAC will operate in synchronous mode and when playing 48K, 96K and 192K, the DAC will operate in asynchronous mode.
You could manually switch the clock lines but this is not only impractical, but it could also upset the DAC requiring a reset.
Easy bypass of USB power. CAUTION: Maximum input voltage for the local regulators is 5.5V
According to the manufacturer, the design employs solid ground plane (a must for high speed digital circuit) and no vias in active circuit (via inductance always create jitter problem). This is apparent from the photo below and in comparison with other designs. The overall layout is very clean and compact.
The datasheet for the Xpresso clocks is actually very extensive, and the jitter measurement is equally extensive. The phase noise plot is included. I have shown here the comparison with the Crystek CCHD-950-80 MHz which is the oscillator I have in my version of the Buffalo II DAC:
Using the 62,5 MHz curve, we get a phase jitter value of 6 psec RMS (10 Hz – 1 MHz). The Crystek CCHD-950-80MHz has a jitter value of about 2 psec RMS. However, at the lower offset frequencies (the ones of interest to audio performance) the phase noise is not very different from that of the Crystek clock which is a good thing.
INTERFACING WITH BII DAC
There is something “a bit odd” with this interface: According to the specifications:
BLCK is the same for both 44/48K and 88/96K. This means that for 44/48K sample rate, the data is running at 128fs and for the rest of the sample rates, it is running at 64fs.
Even though the Sabre32 DAC specifies BCLK to be 64 fs, it appears that it also supports 128fs. Why? because when I play 44.1KHz material, the DAC reports 88.2KHz sample rate and it sounds perfectly fine. (The sample rate reported by the Sabre32 DAC is based on the frequency of the bitclock, the DAC reports 88.2KHz for 44.1K material)
The sample rate for the higher sample rates are reported correctly by the ES9018 DAC.
Is this a “common” feature supported by other DACs?
The AK 4399 supports both a 64fs and a 128fs bitclock:
The Wolfson 8804 also supports 128fs Bitclock -in DSP mode (p. 44):
So it seems that a 128fs bitclock is not so strange after all. I am not sure if any other dacs support 128fs bitclock, but the Sabre32 DAC definitely does.
OTHER NOTEWORTHY IMPLEMENTATIONS
C-Media has been in the computer audio business since its inception. According to their website, their world’s first accomplishment are all related to audio. Its claim to fame is probably the ASUS XONAR series of PC audio cards and interfaces. The latest ASUS XONAR ESSENCE ONE uses the CM6631 part as shown here (There is a review of the Essence One here: [link]):
The Schiit DACs also use the CM6631 part for their USB option board. As discussed here [link] it does not support 176K material because of hardware limitations in the 6631 part. The “A” part supports 176K sample rate.
The Emotive XDA-2 reportedly also uses a CM6631 chip for the USB interface [link]
The MHDT USB Bridge also uses the CM6631 chip [link]
Have yet to see any commercial implementation using the CM6631A part.
diyinhk turns his designs very quickly. This version of the board has been replaced with a new version having isolated I2S [link]
MAC USB 2.0 COMPATIBILITY
There are some reports and fixes to ensure USB2 compatibility with Macintosh computers: [link] with certain CM6631-based implementations.
This board does not suffer from the reported problem: there are no 3-pin devices near pin 98 of the chip.
Plugin this device into a Macintosh computer shows that it is indeed a high speed USB 2.0 device as shown in this snapshot of the CMEDIA device under the USB description screen
JITTER IMMUNITY TEST
Here is a reported test of the CM6631A (implemented in a different device) showing jitter measurements with and without processor load and comparing the asynchronous nature of the USB communication vs a device using USB adaptive communication. [link]
In his conclusion, this device (as well as other USB-asyncrhonous devices) show great immunity against processor loading:
Bottom Line: Don’t worry about jitter! It’s more than likely inaudible in a modern computer system and with decent (not necessarily expensive) audio gear. I see no evidence that high CPU/GPU load makes any difference to jitter. Isolating your DAC from electrical noise polluting the analogue output seems much more important.
I got these photos almost a year ago and they were “lost” in my inbox.
This one uses a yellow LCD. Actually pretty good looking. The quality and finish is of these diy projects is amazing!
The ATC’s (speakers) are custom 150 ASL anniversaries with discrete active amp packs but the room is temporary and centre is coming off the floor asap but it currently sounds nothing short of awesome with the Buff2’s driving the amps direct using legato 3.1.
The PC can decode and output blu ray and full DTS MA HD multi-channel thanks to J River Media Centre which is a superb piece of software.
Getting multi-channel remote volume control without compromising quality was a problem until your code. Next challenge for me will be more buffalo to use the pc for FIR active crossover for the ATC’s…
Dual Mono Buffalo II
The Arduino is well positioned having the USB connector easily accessible for firmware upgrade.