Fundamentals of USB-Audio
I N TH I S D OCU MENT
USB basics
USB-Audio
What’s a second between friends?
Multiple clock sources
Compliance and native support
Summary
USB, the Universal Serial Bus, has been around for decades and is a heavily used
standard in the world of personal computers. Memory sticks, external drives,
mice, and web cameras are all interfaced over USB. In this article we will look into
USB-Audio: a standard for digital audio used in PCs, smart phones and tablets to
interface with audio peripherals such as speakers, microphones, or mixing desks.
In this article we set out to show how USB-Audio works, what to watch out for, and
how to use USB-Audio for high fidelity multi channel input and output.
1
USB basics
USB is a protocol where the PC, the
USB-host,
initiates a
transfer,
and the
device
(for example a USB speaker) responds. Each transfer is addressed to a specific
device, and to a specific
endpoint
on the device.
IN-transfers
send data to the
PC. When the host initiates an IN-transfer the device has to respond with data
for the host. OUT-transfers send data to the device. When the host performs an
OUT-transfer
it sends a packet of data that the device must capture. In the world
of USB-Audio, IN and OUT transfers may be used to transport audio samples: an
OUT-transfer to send audio data from a PC to a speaker, whereas an IN-transfer is
used to send audio data from a microphone to the PC.
There are four sorts of IN and OUT-transfers in USB:
Bulk, Isochronous, Interrupt,
and
Control
transfers.
A bulk transfer is used to
reliably
transfer data between host and device. All USB
transfers carry a CRC (checksum) that indicates whether an error has occurred. On a
bulk transfer, the receiver of the data has to verify the CRC. If the CRC is correct the
transfer is acknowledged, and the data is assumed to have been transferred error-
free. If the CRC is not correct, the transfer is not acknowledged and will be retried. If
the device is not ready to accept data it can send a negative-acknowledgment,
NAK,
which will cause the host to retry the transfer. Bulk transfers are not considered
time criticial, and are scheduled around the time critical transfers discussed below.
Isochronous transfers are used to transfer data in
real-time
between host and
device. When an isochronous endpoint is set up by the host, the host allocates
Publication Date: 2015/3/17
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Document Number: XM007723A
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a specific amount of bandwidth to the isochronous endpoint, and it regularly
performs an IN- or OUT-transfer on that endpoint. For example, the host may OUT
1 KByte of data every 125 us to the device. Since a fixed and limited amount of
bandwidth has been allocated, there is no time to resend data if anything goes
wrong. The data has a CRC as normal, but if the receiving side detects an error
there is no resend mechanism.
Interrupt transfers are used by the host to regularly poll the device to find out
whether something worthwhile has happened. For example, a host may poll an
audio device to check whether the MUTE button has been pressed. The name
Inter-
rupt
transfer is slightly confusing, since they do not interrupt anything. However,
regular polling of data gives the same sort of functionality that an host-interrupt
would provide.
Control transfers are very much like bulk transfers. Control transfers are ac-
knowledged, can be NAKed, and are delivered in a non-real-time fashion. Control
transfers are used for operations that are outside normal data flow, such as query-
ing the device capabilities, or endpoint status. An explanation on how device
capabilities are described is outside the scope of this article, and we just state that
there are predefined
classes
such as ‘USB Audio Class’ or ‘USB Mass Storage Class’
that enable cross platform interoperability.
All transfers are made in USB
frames.
High Speed USB frames span 125 us (Full
Speed USB are 1 ms) and are marked by the host sending a Start-Of-Frame (SOF)
message. Isochronous and Interrupt transfers are transmitted at most once a
frame.
2
USB-Audio
USB-Audio uses isochronous, interrupt and control transfers. All audio data is
transferred over isochronous transfers; interrupt transfers are used to relay infor-
mation regarding the availability of audio clocks; control transfers are used used
to set volume, request sample rates, etc. These are shown in Figure 1.
Figure 1:
Transfers
between a
host and a
USB device:
Isochronous
IN and OUT
for
audio-data,
Control for
setting
parameters,
and Interrupt
for status
monitoring.
Isochronous
OUT transfer
Control
transfer
USB Host (PC)
Interrupt
transfer
Isochronous
IN transfer
USB Device
DAC
Volume
Mute button
MUTE
ADC
Microphone
Speaker
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The data requirements of a USB-Audio system depends on the number of
channels,
the number of
bits
to represent each sample, and the
sample rate.
Typical channel
counts are 2 (stereo), 6 (5.1) or much higher for studio and DJ use. Typically
sample size is 24 bits, although 16 bits is available for legacy audio, and 32 bits
for high quality audio. Typical sample rates are 44.1, 48, 96, and 192 kHz. The
latter is used for high quality audio.
Suppose that we design a stereo audio speaker with a 96 kHz sample rate and
24-bit samples. In order to simplify data marshalling on host and device, 24-bit
values are typically padded with a zero byte, so the total data throughput is 96,000
x 2 channels x 4 bytes = 768,000 bytes per second. The isochronous endpoints
run at a rate of one transfer per 125 us; or 8,000 transfers per second. Dividing
the required byte rate over the frame rate gives us the number of bytes for each
isochronous transfer: 768,000/8,000 = 96 bytes per transfer.
When using CD rates, such as 44,100 Hz, the transfer rate works out as 44.1
transfers per second. In USB-Audio each transfer always carries a whole number of
samples; alternating transfers carry 48 and 40 bytes (6 and 5 stereo samples), so
that the average rate works out as 44.1 bytes per transfer.
A single isochronous transfer can carry 1024 bytes, and can carry at most 256
samples (at 24/32 bits). This means that a single isochronous endpoint can transfer
42 channels at 48 kHz, or 10 channels at 192 kHz (assuming that High Speed USB
is used - Full Speed USB cannot carry more than a single stereo IN and OUT pair at
48 kHz).
When transmitting digital audio, latency is introduced. In the case of High Speed
USB this latency is 250 us. A packet of data is transferred once in every 125 us
window, but given that it may be sent anytime in this window a 250 us buffer is
required. On top of this 250 us delay, extra delay may be incurred in the O/S
driver, and in the CODEC. Note that Full Speed USB has a much higher intrinsic
latency of 2 ms, as data is only sent once in every 1 ms window.
3
What’s a second between friends?
The big issue in digital audio is to agree on a common notion of time. Above we
have defined USB frames to be transferred 8,000 times per second, and set the
speakers to play a sample 96,000 times per second. This will only work if the
speaker and the host agree on the length of a second. USB-Audio offers three
modes that ensure that the host and the speaker agree on timings:
In
synchronous mode,
the length of a second is defined by the host device. That
is, the host will send data at a rate, and the device has to exactly match that
rate.
In
asynchronous mode
it is the other way around, the device sets the definition
of a second, and the host has to match the device.
In
adaptive mode
the data flow determines the clock.
Adaptive and synchronous mode are not ideal because PCs are notoriously bad at
keeping a stable clock, and there are often other audio sources involved, such as
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an external digital deck. Asynchronous mode enables external clock sources to be
used as the master, or a low-jitter clock in the device. Typically, either relies on a
crystal based PLL, as shown in Figure 2.
Figure 2:
A USB-audio
board, with a
crystal for
stable audio
frequencies,
and a
low-jitter PLL
to generate
any
frequency
required.
Hence there are at least two separate clocks in the system, the USB clock with a
host driven frequency of 8,000 transfers per second, and a sample clock with an
externally driven sample rate of, for example, 96,000 Hz.
These clocks will have subtly different frequencies, and the difference will vary
slightly over time. Hence the average number of audio samples per frames will
be slightly more or less than the expected rate. For example, in the case of our
96,000 Hz sample rate, the average number of samples may be 12.001. In order
to ensure that the host sends the right amount of data, and not too much or too
little, the host requests the current sample rate over an interrupt endpoint. Every
few milliseconds the average sample rate over the last period is reported back as a
16.16 bit fixed point number. If the last period averaged out as 12.001 frames,
then the value 0x000C0041 will be reported (65536 * 12.001).
Given this average rate, the host can work out when to send an extra sample in a
transfer; in this example 8 transfers each second will carry one extra sample. In
addition to this, the host can use this value to synchronise itself with the audio
device. This enables host applications such as a DVD player to keep the video in
sync with the audio. If it didn’t, the audio would slowly run ahead of the video, and
after two hours the audio would be a second out.
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In order to keep a short feedback loop, the trick is to not buffer audio packets and
feedback packets unnecessarily. Any additional buffering creates latency in the
reporting, and this latency makes it more difficult to keep a smooth flow of traffic.
This means that the low-level USB stack and the USB-Audio stack should be tightly
integrated, without buffering in between. Although this is hard to achieve on an
application processor, this is quite easy to achieve if the software is implemented
on an embedded processor that has a predictable execution time.
4
Multiple clock sources
The above scheme considers just two clock sources - either the USB device provides
the clock, or the host provides the clock. In more complex devices such as mixer
desks there may be other devices that provide the sample rate, for example through
a digital interface such as ADAT or SPDIF, or through a BNC connector that carries
the word clock. For systems like this, the USB-Audio standard allows designers to
put a
clock selector
in the device.
The clock selector states which clock is to be used as the sample rate. The clock
selector has multiple input clocks (e.g., the incoming clock on an S/PDIF connection;
a local crystal, and the incoming clock on an ADAT connection) and with a control
transfer the user selects which clock to use as input, for example the incoming
clock of the S/PDIF connection.
5
Compliance and native support
Once a device is USB-Audio Class compliant, it will integrate neatly into the oper-
ating system. Figure 3 shows a screenshot of the controls of a USB-Audio device
plugged into Mac OS/X. It shows that the clock selection, sample rate selection,
channel volume control, and mute control are all controllable just like for any other
audio device.
Compliance to the standard makes the device interoperable. O/S vendors can
supply a single USB-Audio driver that drives a multitude of devices, with a multitude
of capabilities.
Indeed, the same USB-Audio implementation can be parameterised to implement a
different number of channels, and the same driver can be used to interface to the
device.
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