Positive Feedback ISSUE 60
march/april 2012

 

DoP open Standard: Method for transferring DSD Audio over PCM Frames Version 1.1
by Andreas Koch

 

The USB Audio specification 2.0 defines multiple formats for audio, of which standard PCM is only one. A general "raw data" format was also defined that can be used for any kind of data including audio, but unfortunately, no specific format was defined for DSD. With the ongoing proliferation of USB converters in the current market, it appears that the opportunity for the official USB specification to adopt a single common method of transferring DSD audio via USB is slowly disappearing. This article is an attempt at uniting as many manufacturers as possible and jointly defining a method for transferring DSD via USB. [For the first article in this series on DSD over USB, please see PF Issue 59 at http://www.positive-feedback.com/Issue59/dsd2.htm.] While this method is mainly targeted for USB links, it is general enough to be applied to other PCM-based links such as Firewire, AES/EBU, S/PDIF, etc.

1. Motivation

The manufacturers developing audio playback software want to minimize the number of formats they need to support for a USB link. Ideally there is only a single such format. Likewise hardware manufacturers want to make their hardware compatible with as many playback platforms as possible. That, of course, only happens when all use the same format.

As mentioned above, USB Audio already supports a "raw data" format that could be used for DSD and that would create a clear separation to any audio data path containing PCM. However, the latest release of Apple’s operating system OS 10.7 incorporates a USB driver that only supports PCM. Furthermore, the central audio engine, CoreAudio, inside the OS only supports PCM as well, though luckily with no limitation on sample rate. (Earlier versions of the Apple OS supported a mode that was compatible with raw data mode, but that is history now.) Since the architecture of Apple’s OS forces audio software developers to use CoreAudio for everything audio related, there is basically only 1 format left for the Mac platform: PCM. Creating a separate path for DSD would involve a lot of surgery, if it is even possible. So we have no choice but to use a PCM path to transfer DSD audio, by using special flags or headers that allow the receiving hardware to detect a format change and switch their decoder accordingly.

When using the Windows platform things are little easier: Windows by nature does not fully support USB Audio 2.0, and what it does support is limited to PCM only at a sample rate of 96kHz or less. There is no native driver support for higher resolution PCM, and it is clear from the beginning that a custom driver needs to be created for this platform, whether it is for regular PCM or DSD. Luckily a 3rd party software developer (Steinberg Audio) jumped in and created a driver (called ASIO) already many years ago that supports PCM and DSD, with no limitation on sample rate or wordlength. It has become quite popular, and many software vendors support this in the meantime. ASIO is not directly a hardware driver, but sits between the audio playback application and the hardware driver. Each hardware manufacturer still needs to develop a custom hardware driver for their own hardware, but ASIO then creates a common interface standard for all application software.

2. Solutions

As seen above, the Windows platform basically offers a solution with the ASIO driver and the raw data format supported by USB Audio 2.0. This is not as ideal as having a dedicated DSD path via USB, but this is safe and straightforward.

Since the Apple OS only allows a PCM path, we have to find a way to put DSD audio data into PCM frames, which are then sent via the native USB driver. DSD has a sample size of 1 bit and a sample rate of 2.8224MHz. In other words, the data rate is 2.8224Mbits/sec. This is equivalent to 16-bit PCM at a rate of 176.4kHz. In order to clearly identify when this PCM stream contains DSD and when it contains PCM, we will need additional bits. The PCM format with the next higher bit rate is 24 bits at a sample rate of 176.4kHz. This gives us 8 extra bits for this marker of identifier. It seems like a bit overkill if all we need is 2 states (8 bits give us 256 states), but we will see that this extra overhead comes in handy. Here is how we can use the 24 bits in each sample and for each channel:

The 8 most significant bits are used for the DSD marker, and alternate with each sample between 0x05 and 0xFA. Each channel within a sample contains the same marker. This has been chosen to minimize the click that might be experienced when the receiving hardware misinterprets the data as PCM when it really is DSD. If this should happen, it would create a tone around 88kHz and roughly -34dB, nothing harmful and something that most D/A converters would suppress to some degree before it even reaches the loudspeaker. It should be pointed out that hardware manufacturers and software developers alike can easily use common safeguards to prevent such cases of erroneous format switching, and that they may only be limited to times during development of hardware and software. It is their responsibility to prevent misinterpreted cases and to test their products thoroughly before release. Misinterpretation of PCM data as DSD may create less predictable clicks.

The remaining 16 lower bits are then used for the DSD data, first or oldest bit in slot t0. The USB Audio specification assigns each PCM Frame to a specific channel (left, right, etc.), and when used for DSD streaming, each PCM Frame contains only DSD data corresponding to its assigned channel.

3. Solutions for double rate DSD (128FS) and beyond

Two solutions are possible depending on whether or not the used PCM transmission scheme is capable of supporting the PCM rate of 352.8kHz:

  • The solution described above for 64FS DSD can easily be extended for 128FS by simply raising the underlying PCM sample rate from 176.4kHz to 352.8kHz. All the marker bytes and bit ordering remain the same.

  • For those conduits that do not support 352.8kHz (such as AES/EBU), an alternative method can be used without raising the PCM sample rate:

A PCM channel pair (for instance L/R) is used to transmit 128FS DSD for a single DSD channel. The lower PCM frame number contains the 16 older DSD bits in the same order as in the 64FS case. The higher PCM frame number in the pair then contains the newer 16 DSD bits. A different marker byte is used to distinguish this method from the first one.

Solution 1 can easily be extended to support even higher DSD rates by raising the underlying PCM rate.

4. Recommended implementation

While there is certainly more than one way to implement this solution on the receiver side, the authors found the following implementation to work reliably:

  • In order to switch from PCM to DSD mode the receiver has to detect 32 consecutive DSD marker bytes on all channels used.

  • In order to switch from DSD to PCM mode the receiver has to detect at least 1 single missing DSD marker byte in at least 1 channel.

This introduces an additional latency of around 180usec. If the USB buffers are accessible for reading while the USB microframe is still being received, then no additional delay is necessary.

If the transmission link used is USB, it is recommended that for 128FS DSD the first solution described in section 3 is used, or else significant bandwidth is wasted for PCM transmission, as the second solution always requires double the amount of channels than is necessary for PCM transmission.

In order to minimize false detection where DSD data would be interpreted as PCM or vice versa, it is recommended that the host software verifies the DSD capability of the hardware before exchanging any data. This can be done in various ways, and depends on physical link, driver, and computer platform.

5. Industry Support

The following have contributed to this document and/or pledged their support for this format (alphabetical order):

Aesthetix Jim White

Audirvana Damien Plisson

CEntrance Michael Goodman

CH-Precision Thierry Heeb, Florian Cossy

ChannelD Rob Robinson

dCS Andy McHarg, David J Steven

JRiver, Inc. Matt Ashland

Light Harmonic Larry Ho

Merging Technologies Dominique Brulhart

MSB Technology Larry S. Gullman

Mytek Digital Michal Jurewicz

Playback Designs Andreas Koch

Signalyst Jussi Laako

Sonic Solutions Jon Reichbach

Wavelength Audio Gordon Rankin

XMOS Ali Dixon

Vitus Audio Martin Kristensen

Independent Dustin Forman

About the Author

Andreas Koch has been involved in the creation of SACD from the beginning while working at Sony. He was leading a team of engineers designing the world's first multichannel DSD recorder and editor for professional recording (Sonoma workstation), world's first multichannel DSD converters (ADC and DAC) and participated in various standardization committees world-wide for SACD. Later he went on as a consultant to design a number of proprietary DSD processing algorithms for converting PCM to DSD and DSD to PCM, and other technologies for D/A conversion and clock jitter control in DACs. In 2008 he co-founded Playback Designs to bring to market his exceptional experience and know-how in DSD in form of D/A converters and CD/SACD players. Earlier he was part of an engineering team at Studer in Switzerland designing one of the world's first digital tape recorders, then lead a team of engineers working on a multichannel hard disk recorder, anda 3-year stint at Dolby as the company's first digital design engineer gave him a well rounded foundation of audio know-how and experience. He can be reached at andreas@akdesigninc.com.

 

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