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|
<html devsite>
<head>
<title>Hearing Aid Audio Support Using Bluetooth LE</title>
<meta name="project_path" value="/_project.yaml" />
<meta name="book_path" value="/_book.yaml" />
</head>
<body>
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<p>
Hearing aid devices (HA) can have improved accessibility on
Android-powered mobile devices by using connection-oriented L2CAP
channels (CoC) over Bluetooth Low Energy (BLE). CoC uses an elastic
buffer of several audio packets to maintain a steady flow of audio, even
in the presence of packet loss. This buffer provides audio quality for
hearing aid devices at the expense of latency.
</p>
<p>
The design of CoC references the
<a href="https://www.bluetooth.com/specifications/bluetooth-core-specification">Bluetooth Core Specification Version 5</a>
(BT). To stay aligned with the core specifications, all multi-byte
values on this page should be read as little-endian.
</p>
<h2 id="terminology">Terminology</h2>
<ul>
<li>
<strong>Central</strong> - the Android device that scans for
advertisements over Bluetooth.
</li>
<li>
<strong>Peripheral</strong> - the hearing instrument that sends
advertisement packets over Bluetooth.
</li>
</ul>
<h2 id="network-topology-and-system-architecture">
Network topology and system architecture
</h2>
<p>
When using CoC for hearing aids, the network topology assumes a single
central and two peripherals, one left and one right, as seen in
<strong>Figure 1</strong>. The Bluetooth audio system views the left
and right peripherals as a single audio sink. If a peripheral is
missing, due to a monaural fit or a loss of connection, then the
central mixes the left and right audio channel and transmits the audio
to the remaining peripheral. If the central loses connection to both
peripherals, then the central considers the link to the audio sink
lost. In those cases, the central routes audio to another output.
</p>
<p><img src="/devices/bluetooth/images/bt_asha_topology.png"><br />
<strong>Figure 1.</strong> Topology for pairing hearing aids with
Android mobile devices using CoC over BLE
</p>
<p>
When the central is not streaming audio data to the peripheral and can
maintain a BLE connection, the central should not disconnect from the
peripheral. Maintaining the connection allows the data communication
to the GATT server residing on the peripheral.
</p>
<aside class="note">
<strong>Note</strong>: There is no audio backlink between the central
and the peripherals. During a phone call the central microphones are
used for voice input.
</aside>
<p>
When pairing and connecting hearing devices, the central should:
</p>
<ul>
<li>
Keep track of the more recent left and right peripherals paired.
Those two peripherals should be considered the audio sink.
</li>
<li>
Assume the peripherals are in use if a valid pairing exists. The
central should attempt to connect or reconnect with the paired
device when the connection is lost.
</li>
<li>
Assume the peripherals are no longer in use if a pairing is deleted.
</li>
</ul>
<p>
In the cases above, <em>pairing</em> refers to the action of
registering a set of hearing aids with a given UUID and
left/right designators in the OS, not the Bluetooth pairing process.
</p>
<h2 id="system-requirements">System requirements</h2>
<p>
To properly implement CoC for a good user experience, the Bluetooth
systems in the central and peripheral devices should:
</p>
<ul>
<li>
implement a compliant BT 4.2 or higher controller
</li>
<li>
support at least 2 simultaneous LE links with parameters as
described in <a href="#audio-packet-format-and-timing">Audio packet
format and timing</a>.
</li>
<li>
have an LE credit based flow control [BT Vol 3, Part A, Sec 10.1].
Devices should support an MTU and MPS size of at least 240 bytes on
CoC and be able to buffer up to 8 packets.
</li>
<li>
have an LE data length extension [BT Vol 6, Part B, Sec 5.1.9] with
a payload of at least 87 bytes. It is recommended that the data
length is at least 250 bytes.
</li>
<li>
have the central device support the HCI LE Connection Update Command
and comply with the non-zero minimum_CE_Length parameter.
</li>
<li>
maintain the data throughput for two LE CoC connections to two
different peripherals with the connection intervals and payload
sizes in <a href="#audio-packet-format-and-timing">Audio packet
format and timing</a>.
</li>
<li>
have the peripheral set the <code>MaxRxOctets</code> and
<code>MaxRxTime</code> parameters in the <code>LL_LENGTH_REQ</code>
or <code>LL_LENGTH_RSP</code> frames to be the smallest required values
that are necessary for these specifications. This lets the central
optimize its time scheduler when calculating the amount of time
needed to receive a frame.
</li>
</ul>
<p>
The peripheral and central may implement 2 Mbit PHY as specified in
BT 5. The central should support audio links up to 64 kbit/s on both 1
Mbit and 2 Mbit PHY but can choose to limit support for links
requiring more than 64 kbit/s to the 2 Mbit PHY in order to improve
coexistence with other 2.4 GHz devices. The BLE long range PHY should
not be used.
</p>
<p>
CoC uses the standard Bluetooth mechanisms for link layer encryption
and frequency hopping.
</p>
<h2 id="asha-gatt-services">ASHA GATT services</h2>
<p>
A peripheral should implement the Audio Streaming for Hearing Aid
(ASHA) GATT server service described below. The peripheral should
advertise this service when in general discoverable mode to let the
central recognize an audio sink. Any LE audio streaming operations
shall require encryption. The BLE audio streaming consists of the
following characteristics:
</p>
<table>
<tr>
<th>Characteristic</th>
<th>Properties</th>
<th>Description</th>
</tr>
<tr>
<td>ReadOnlyProperties</td>
<td>Read</td>
<td>See <a href="#readonlyproperties">ReadOnlyProperties</a>.</td>
</tr>
<tr>
<td>AudioControlPoint</td>
<td>Write without Response</td>
<td>
Control point for audio stream. See
<a href="#audiocontrolpoint">AudioControlPoint</a>.
</td>
</tr>
<tr>
<td>AudioStatusPoint</td>
<td>Read/Notify</td>
<td>
Status report field for the audio control point. Opcodes are:
<ul>
<li><strong>0</strong> - Status OK</li>
<li><strong>-1</strong> - Unknown command</li>
<li><strong>-2</strong> - Illegal parameters</li>
</ul>
</td>
</tr>
<tr>
<td>Volume</td>
<td>Write without Response</td>
<td>
Byte between -128 and 0 indicating volume in dB. -128 should be
interpreted as mute. 0 dB with a rail-to-rail sine tone streamed
should represent a 100 dBSPL input equivalent on the hearing
instrument. The central should stream in nominal full scale and
use this variable to set the desired presentation level in the
peripheral.
</td>
</tr>
<tr>
<td>LE_PSM</td>
<td>Read</td>
<td>
PSM to use for connecting the audio channel. To be picked from the
dynamic range [BT Vol 3, Part A, Sec 4.22]
</td>
</tr>
</table>
<p>The UUIDs assigned to the service and characteristics:</p>
<p><strong>Service UUID</strong>: <code>{0xFDF0}</code></p>
<table>
<tr>
<th>Characteristic</th>
<th>UUID</th>
</tr>
<tr>
<td>ReadOnlyProperties</td>
<td><code>{6333651e-c481-4a3e-9169-7c902aad37bb}</code></td>
</tr>
<tr>
<td>AudioControlPoint</td>
<td><code>{f0d4de7e-4a88-476c-9d9f-1937b0996cc0}</code></td>
</tr>
<tr>
<td>AudioStatus</td>
<td><code>{38663f1a-e711-4cac-b641-326b56404837}</code></td>
</tr>
<tr>
<td>Volume</td>
<td><code>{00e4ca9e-ab14-41e4-8823-f9e70c7e91df}</code></td>
</tr>
<tr>
<td>LE_PSM</td>
<td><code>{2d410339-82b6-42aa-b34e-e2e01df8cc1a}</code></td>
</tr>
</table>
<p>
In addition to the ASHA GATT service, the peripheral should also
implement the Device Information Service to let the central detect the
manufacturer names and device names of the peripheral.
</p>
<h3 id="readonlyproperties">ReadOnlyProperties</h3>
<p>ReadOnlyProperties have the following values:</p>
<table>
<tr>
<th>Byte</th>
<th>Description</th>
</tr>
<tr>
<td>0</td>
<td>Version - must be 0x01</td>
</tr>
<tr>
<td>1</td>
<td>See <a href="#devicecapabilities">DeviceCapabilities</a>.</td>
</tr>
<tr>
<td>2-9</td>
<td>See <a href="#hisyncid">HiSyncId</a>.</td>
</tr>
<tr>
<td>10</td>
<td>See <a href="#featuremap">FeatureMap</a><strong>.</strong></td>
</tr>
<tr>
<td>11-12</td>
<td>
RenderDelay. This is the time, in milliseconds, from when the
peripheral receives an audio frame until the peripheral renders
the output. These bytes can be used to delay a video to
synchronize with the audio.
</td>
</tr>
<tr>
<td>13-14</td>
<td>
PreparationDelay. This is the time, in milliseconds, the
peripheral needs in order to render audio after the start
command has been issued,such as for loading codecs. The
PreparationDelay can be used by the central to delay audio
playback of short messages.
</td>
</tr>
<tr>
<td>15-16</td>
<td>
Supported <a href="#codec-ids">Codec IDs</a>. This is a bitmask
of supported codec IDs. A 1 in a bit location corresponds to a
supported codec. All other bits should be set to 0.
</td>
</tr>
</table>
<h4 id="devicecapabilities">DeviceCapabilities</h4>
<table>
<tr>
<th>Bit</th>
<th>Description</th>
</tr>
<tr>
<td>0</td>
<td>Device side (Left: 0, Right: 1).</td>
</tr>
<tr>
<td>1</td>
<td>
Monaural (0) / Binaural (1). Indicates whether the device is
stand-alone and receives mono data, or if the device is part
of a set.
</td>
</tr>
<tr>
<td>2-7</td>
<td>Reserved (set to 0).</td>
</tr>
</table>
<h4 id="hisyncid">HiSyncID</h4>
<table>
<tr>
<th>Byte</th>
<th>Description</th>
</tr>
<tr>
<td>0-1</td>
<td>ID of the manufacturer.</td>
</tr>
<tr>
<td>2-7</td>
<td>
Unique ID identifying the hearing aid set. This ID must be set
to the same on both the left and the right peripheral.
</td>
</tr>
</table>
<h4 id="featuremap">FeatureMap</h4>
<table>
<tr>
<th>Bit</th>
<th>Description</th>
</tr>
<tr>
<td>0</td>
<td>LE CoC audio streaming supported (Yes/No).</td>
</tr>
<tr>
<td>1-7</td>
<td>Reserved (set to 0).</td>
</tr>
</table>
<h4 id="codec-ids">Codec IDs</h4>
<p>
If the bit is set, then that particular codec is support.
</p>
<table>
<tr>
<th>Bit number</th>
<th>Codec and sample rate</th>
<th>Required bitrate</th>
<th>Frame time</th>
<th>Mandatory on central (C) or peripheral (P)</th>
</tr>
<tr>
<td>0</td>
<td>Reserved</td>
<td>Reserved</td>
<td>Reserved</td>
<td>Reserved</td>
</tr>
<tr>
<td>1</td>
<td>G.722 @ 16 kHz</td>
<td>64 kbit/s</td>
<td>Variable</td>
<td>C and P</td>
</tr>
<tr>
<td>2</td>
<td>G.722 @ 24 kHz</td>
<td>96 kbit/s</td>
<td>Variable</td>
<td>C</td>
</tr>
<tr>
<tr>
<td colspan="5">
3-15 are reserved.<br />
0 is also reserved.
</td>
</tr>
</table>
<h3 id="audiocontrolpoint">AudioControlPoint</h3>
<p>
This control point cannot be used when the LE CoC is closed. See
<a href="#starting-and-stopping-an-audio-stream">Starting and
stopping an audio stream</a> for the procedure description.
</p>
<table>
<tr>
<th>Opcode</th>
<th>Arguments</th>
<th>Description</th>
</tr>
<tr>
<td>1 <code>«Start»</code></td>
<td>
<ul>
<li><code>uint8_t codec</code></li>
<li><code>uint8_t audiotype</code></li>
<li><code>int8_t volume</code></li>
</ul>
</td>
<td>
Instructs the peripheral to reset the codec and start the
playback of frame 0. The codec field indicates the bit number
of the Codec ID to use for this playback.<br /><br />
The audio type bit field indicates the audio type(s) present
in the stream:
<ul>
<li><strong>0</strong> - Unknown</li>
<li><strong>1</strong> - Ringtone</li>
<li><strong>2</strong> - Phonecall</li>
<li><strong>3</strong> - Media</li>
</ul>
The peripheral should not request connection updates before a
<code>«Stop»</code> opcode has been received.
</td>
</tr>
<tr>
<td>2 <code>«Stop»</code></td>
<td>None</td>
<td>
Instructs the peripheral to stop rendering audio. A new audio
setup sequence should be initiated following this stop in order
to render audio again. The peripheral may request a connection
update following this command.
</td>
</tr>
</table>
<h2 id="advertisements-for-asha-gatt-service">
Advertisements for ASHA GATT Service
</h2>
<p>
The <a href="#asha-gatt-services">service UUID</a> must be in the
advertisement packet. In either the advertisement or the scan
response frame, the peripherals must have a Service Data:
</p>
<table>
<tr>
<th>Byte offset</th>
<th>Name</th>
<th>Description</th>
</tr>
<tr>
<td>0</td>
<td>AD Length</td>
<td>>= 0x09</td>
</tr>
<tr>
<td>1</td>
<td>AD Type</td>
<td>0x16 (Service Data - 16-bits UUID)</td>
</tr>
<tr>
<td>2-3</td>
<td>Service UUID</td>
<td>
0xFDF0 (little-endian)<br /><br />
<strong>Note:</strong> This is a temporary ID.
</td>
</tr>
<tr>
<td>4</td>
<td>Protocol Version</td>
<td>0x01</td>
</tr>
<tr>
<td>5</td>
<td>Capability</td>
<td>
<ul>
<li><strong>0</strong> - left (0) or right (1) side</li>
<li><strong>1</strong> - single (0) or dual (1) devices.</li>
<li>
<strong>2-7</strong> - reserved. These bits must be zero.
</li>
</ul>
</td>
</tr>
<tr>
<td>6-9</td>
<td>Truncated <a href="#hisyncid">HiSyncID</a></td>
<td>
Four least significant bytes of the
<a href="#hisyncid">HiSyncId</a>.
</td>
</tr>
</table>
<p>
The peripherals must have a <strong>Complete Local Name</strong>
data type that indicates the name of the hearing aid. This name will
be used on the mobile device's user interface so the user can select
the right device. The name should not indicate the left or right
channel since this information is provided in
<a href="#devicecapabilities">DeviceCapabilities</a>.
</p>
<p>
If the peripherals put the name and ASHA service data types in the same
frame type (ADV or SCAN RESP), then the two data types should appear
in the same frame. This lets the mobile device scanner get both data
in the same scan result.
</p>
<p>
During the initial pairing, it is important that the peripherals
advertise at a rate fast enough to let the mobile device quickly
discover the peripherals and bond to them.
</p>
<h2 id="synchronizing-left-and-right-peripheral-devices">Synchronizing left and right peripheral devices</h2>
<p>
To work with Bluetooth on Android mobile devices, peripheral devices
are responsible for ensuring that they are synchronized. The playback
on the left and right peripheral devices needs to be synchronized in
time. Both peripheral devices must play back audio samples from the
source at the same time.
</p>
<p>
Peripheral devices can synchronize their time by using a sequence
number prepended to each packet of the audio payload. The central will
guarantee that audio packets that are meant to be played at the same
time on each peripheral have the same sequence number. The sequence
number is incremented by one after each audio packet. Each sequence
number is 8-bit long, so the sequence numbers will repeat after 256
audio packets. Since each audio packet size and sample rate is fixed
for each connection, the two peripherals can deduce the relative
playing time. For more information about the audio packet, see
<a href="#audio-packet-format-and-timing">Audio packet format and
timing</a>.
</p>
<h2 id="audio-packet-format-and-timing">Audio packet format and timing</h2>
<p>
Packing audio frames (blocks of samples) into packets lets the hearing
instrument derive timing from the link layer timing anchors. To
simplify the implementation:
</p>
<ul>
<li>
An audio frame should always match the connection interval in time.
For example, if the connection interval is 15ms and sample rate is
1 kHz, then the audio frame should contain 240 samples.
</li>
<li>
Sample rates in the system are restricted to multiples of 8kHz to
always have an integer number of samples in a frame regardless of
the frame time or the connection interval.
</li>
<li>
A sequence byte should prepend audio frames. The sequence byte
should be counting with wrap-around and allow the peripheral to
detect buffer mismatch or underflow.
</li>
<li>
An audio frame should always fit into a single LE packet. The audio
frame should be sent as a separate L2CAP packet. The size of the LE
LL PDU should be:<br />
<em>audio payload size + 1 (sequence counter) + 6
(4 for L2CAP header, 2 for SDU)</em>
</li>
<li>
A connection event should always be large enough to contain 2 audio
packets and 2 empty packets for an ACK to reserve bandwidth for
retransmissions.
</li>
</ul>
<p>
To give the central some flexibility, the G.722 packet length is not
specified. The G.722 packet length can change based on the connection
interval that the central sets.
</p>
<p>
For all the codecs that a peripheral supports, the peripheral should
support the connection parameters below. This is a non-exhaustive list
of configurations that the central can implement.
</p>
<table>
<tr>
<th>Codec</th>
<th>Bitrate</th>
<th>Connection interval</th>
<th>CE Length (1/2 Mbit)</th>
<th>Audio payload size</th>
</tr>
<tr>
<td>G.722 @ 16 kHz</td>
<td>64 kbit/s</td>
<td>10 ms</td>
<td>2500 / 2500 us</td>
<td>80 bytes</td>
</tr>
<tr>
<td>G.722 @ 16 kHz</td>
<td>64 kbit/s</td>
<td>20 ms</td>
<td>5000/3750 us</td>
<td>160 bytes</td>
</tr>
<tr>
<td>G.722 @ 24 kHz</td>
<td>96 kbit/s</td>
<td>10 ms</td>
<td>3750 / 2500 us</td>
<td>120 bytes</td>
</tr>
<tr>
<td>G.722 @ 24 kHz</td>
<td>96 kbit/s</td>
<td>20 ms</td>
<td>5000 / 3750 us</td>
<td>240 bytes</td>
</tr>
</table>
<h2 id="starting-and-stopping-an-audio-stream">
Starting and stopping an audio stream
</h2>
<aside class="note">
<strong>Note:</strong> This section is based on simulations with an
audio frame buffer depth of 6. This depth is enough to prevent
underflow on the peripheral in most packet loss scenarios. With this
depth, the network delay in the system is six times the connection
interval. To keep delays down, a short connection interval is
preferred.
</aside>
<p>
Before starting an audio stream, the central queries the peripherals
and establishes the best quality common denominator codec. The stream
setup then proceeds through sequence:
</p>
<ol>
<li>PSM, and optionally, PreparationDelay is read.</li>
<li>
CoC L2CAP channel is opened – the peripheral should grant 8 credits
initially.
</li>
<li>
A connection update is issued to switch the link to the parameters
required for the chosen codec.
</li>
<li>
Both the central and the peripheral host wait for the update
complete event.
</li>
<li>
Restart the audio encoder, and reset the packet sequence count to 0.
A <code>«Start»</code> command with the relevant parameters is
issued on the AudioControlPoint. During audio streaming, the replica
should be available at every connection event even though the current
replica latency may be non-zero.
</li>
<li>
The peripheral takes the first audio packet from its internal queue
(sequence number 0) and plays it.
</li>
</ol>
<p>
The central issues the <strong>«Stop»</strong> command to close the
audio stream. Once the audio stream is closed, the peripheral may ask
for more relaxed connection parameters. Go through the sequence above
again to restart the audio streaming. When the central is not
streaming audio, it should still maintain a LE connection for GATT
services.
</p>
<p>
The peripheral should not issue a connection update to the central.
To save power, the central may issue a connection update to the
peripheral when it is not streaming audio.
</p>
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