U.S. patent application number 15/842107 was filed with the patent office on 2018-05-17 for method for streaming through a data service over a radio link subsystem.
The applicant listed for this patent is Ibiquity Digital Corporation. Invention is credited to Russell Iannuzzelli.
Application Number | 20180139499 15/842107 |
Document ID | / |
Family ID | 40754415 |
Filed Date | 2018-05-17 |
United States Patent
Application |
20180139499 |
Kind Code |
A1 |
Iannuzzelli; Russell |
May 17, 2018 |
METHOD FOR STREAMING THROUGH A DATA SERVICE OVER A RADIO LINK
SUBSYSTEM
Abstract
An apparatus for controlling a data rate in a data client for a
digital audio broadcasting system includes a buffer for storing
data, a codec for coding data, and a control module for controlling
a bit rate of the codec in response to a level of the data in the
buffer. A method performed by the apparatus is also included.
Inventors: |
Iannuzzelli; Russell;
(Bethesda, MD) |
|
Applicant: |
Name |
City |
State |
Country |
Type |
Ibiquity Digital Corporation |
Columbia |
MD |
US |
|
|
Family ID: |
40754415 |
Appl. No.: |
15/842107 |
Filed: |
December 14, 2017 |
Related U.S. Patent Documents
|
|
|
|
|
|
Application
Number |
Filing Date |
Patent Number |
|
|
11958783 |
Dec 18, 2007 |
9872066 |
|
|
15842107 |
|
|
|
|
Current U.S.
Class: |
1/1 |
Current CPC
Class: |
H04H 60/07 20130101;
H04N 21/23406 20130101; H04N 21/4392 20130101; H04H 2201/183
20130101; H04H 2201/186 20130101; H04N 21/44004 20130101 |
International
Class: |
H04N 21/44 20060101
H04N021/44; H04N 21/439 20060101 H04N021/439 |
Claims
1.-18. (canceled)
19. A radio signal broadcasting system, the system comprising: a
radio frequency (RF) transmitter operatively configured to
broadcast an in-band on-channel (IBOC) analog broadcast signal via
an RF channel, and stream video data via at least one sub-channel
of the first RF channel; a studio transmitter link (STL) to the RF
transmitter, the STL including: a buffer configured to store
service data for streaming, including the video data, via the at
least one sub-channel of the RF channel; a codec configured for
encoding the service data for streaming; and a control module
configured for controlling a bit rate of the codec in response to a
fill level of the service data in the buffer.
20. The system of claim 19, wherein the control module is
configured to change the bit rate of the codec according to the
fill level of the service data in the buffer, wherein the change in
bit rate is phase continuous.
21. The system of claim 19, wherein the control module is
configured to compare the level of the service data in the buffer
to a first threshold level and reduce the bit rate of the codec
when the level of the service data in the buffer exceeds the first
threshold level.
22. The system of claim 21, wherein the control module is further
configured to compare the level of the service data in the buffer
to a second threshold level and increase the bit rate of the codec
when the level of the service data in the buffer is below the
second threshold level.
23. The system of claim 19, wherein the control module is
configured to compare the level of the service data in the buffer
to a second threshold level and suspend requests for service data
when the level of the service data in the buffer is below the
second threshold level.
24. The system of claim 19, wherein the STL includes an importer
operatively coupled to the codec, wherein the buffer receives the
service data at a first data rate and the importer requests service
data from the buffer at a second data rate, and wherein the second
data rate is determined according to a transmit data rate of the at
least one sub-channel of the RF channel.
25. The system of claim 24, including an exporter configured to
multiplex the streaming service data with main program audio data
to generate an output signal, wherein the exporter requests data
from the importer at the second data rate.
26. The system of claim 24, wherein the buffer is configured to
receive the video data from a video data source, and wherein access
to the buffer by the service data source is asynchronous to access
to the buffer by the codec.
27. The system of claim 24, wherein the control module compares the
level of the service data in the buffer to a second threshold level
and suspends requests for the service data by the importer when the
level of the service data in the buffer is below the second
threshold level.
28. The system of claim 19, wherein the control module is
configured for controlling a compression rate of the codec in
response to a fill level of the service data in the buffer.
29. A method for controlling a data rate in a radio signal
broadcasting system, the method comprising: broadcasting an in-band
on-channel (IBOC) analog broadcast signal via a radio frequency
(RF) channel using an RF transmitter; streaming service data via at
least one sub-channel of the RF channel simultaneously with the
broadcasting, wherein the service data includes video data;
buffering the service data for the streaming; encoding the buffered
service data for streaming using a codec; and controlling a bit
rate of the codec according to a buffered level of the service
data.
30. The method of claim 29, wherein the controlling the bit rate of
the codec includes changing the bit rate of the codec in response
to a fill level of the service data in a buffer, wherein the change
in bit rate of the codec is phase continuous.
31. The method of claim 29, wherein the controlling the bit rate of
the codec includes comparing the buffered level of the service data
to a first threshold level and decreasing the bit rate of the codec
when the buffered level of the service data exceeds the first
threshold level.
32. The method of claim 31, wherein the controlling the bit rate of
the codec includes comparing the buffered level of the service data
to a second threshold level and increasing the bit rate of the
codec when the buffered level of the service data is less than the
second threshold level.
33. The method of claim 29, wherein the controlling the bit rate of
the codec includes comparing the buffered level of the service data
to a second threshold level and suspending requests for buffered
service data when the level of the service data is below the second
threshold level.
34. The method of claim 29, wherein the buffering the service data
for streaming includes placing the service data in a buffer at a
first data rate, wherein the streaming the service data includes
requesting the service data for streaming from the buffer at a
second data rate different from the first data rate, and wherein
the second data rate is determined according to a transmit data
rate of the at least one sub-channel of the RF channel.
35. The method of claim 29, wherein the streaming the service data
includes multiplexing the streaming service data with digital audio
data to generate multiplexed data in an output signal transmitted
by the RF transmitter.
36. The method of claim 29, wherein the buffering the service data
for the streaming includes receiving the video data from a video
data source for buffering, and wherein encoding the buffered
service data includes accessing the buffered video data
asynchronously to the buffering of the video data.
37. The method of claim 29, wherein the controlling a bit rate of
the codec includes controlling a compression rate of the codec in
response to the buffered level of the service data.
38. The method of claim 29, wherein the broadcasting the IBOC
analog broadcast signal includes broadcasting the IBOC analog
signal at a transmitter site, and wherein the buffering the service
data for streaming includes buffering the service data for
streaming at a studio site remote from the transmitter site.
Description
FIELD OF THE INVENTION
[0001] This invention relates to methods and apparatus for
controlling data, and more particularly, to the application of such
methods and apparatus for streaming in a data subsystem of a
digital audio broadcasting system.
BACKGROUND OF THE INVENTION
[0002] Digital radio broadcasting technology delivers digital audio
and data services to mobile, portable, and fixed receivers. One
type of digital radio broadcasting, referred to as in-band
on-channel (IBOC) digital audio broadcasting (DAB), uses
terrestrial transmitters in the existing Medium Frequency (MF) and
Very High Frequency (VHF) radio bands. HD Radio.TM. technology,
developed by iBiquity Digital Corporation, is one example of an
IBOC implementation for digital radio broadcasting and reception.
IBOC DAB signals can be transmitted in a hybrid format including an
analog modulated carrier in combination with a plurality of
digitally modulated carriers or in an all-digital format wherein
the analog modulated carrier is not used. Using the hybrid mode,
broadcasters may continue to transmit analog AM and FM
simultaneously with higher-quality and more robust digital signals,
allowing themselves and their listeners to convert from
analog-to-digital radio while maintaining their current frequency
allocations.
[0003] One feature of digital transmission systems is the inherent
ability to simultaneously transmit both digitized audio and data.
Thus the technology also allows for wireless data services from AM
and FM radio stations. The broadcast signals can include metadata,
such as the artist, song title, or station call letters. Special
messages about events, traffic, and weather can also be included.
For example, traffic information, weather forecasts, news, and
sports scores can all be scrolled across a radio receiver's display
while the user listens to a radio station.
[0004] The design provides a flexible means of transitioning to a
digital broadcast system by providing three new waveform types:
Hybrid, Extended Hybrid, and All-Digital. The Hybrid and Extended
Hybrid types retain the analog FM signal, while the All-Digital
type does not. All three waveform types conform to the currently
allocated spectral emissions mask.
[0005] The digital signal is modulated using Orthogonal Frequency
Division Multiplexing (OFDM). OFDM is a parallel modulation scheme
in which the data stream modulates a large number of orthogonal
subcarriers, which are transmitted simultaneously. OFDM is
inherently flexible, readily allowing the mapping of logical
channels to different groups of subcarriers.
[0006] The HD Radio system allows multiple services to share the
broadcast capacity of a single station. One feature of digital
transmission systems is the inherent ability to simultaneously
transmit both digitized audio and data. Thus the technology also
allows for wireless data services from AM and FM radio stations.
First generation (core) services include a Main Program Service
(MPS) and the Station Information Service (SIS). Second generation
services, referred to as Advanced Application Services (AAS),
include information services providing, for example, multicast
programming, electronic program guides, navigation maps, traffic
information, multimedia programming and other content. The AAS
Framework provides a common infrastructure to support the
developers of these services. The AAS Framework provides a platform
for a large number of service providers and services for
terrestrial radio. It has opened up numerous opportunities for a
wide range of services (both audio and data) to be deployed through
the system.
[0007] The National Radio Systems Committee, a standard-setting
organization sponsored by the National Association of Broadcasters
and the Consumer Electronics Association, adopted an IBOC standard,
designated NRSC-5A, in September 2005. NRSC-5A, the disclosure of
which is incorporated herein by reference, sets forth the
requirements for broadcasting digital audio and ancillary data over
AM and FM broadcast channels. The standard and its reference
documents contain detailed explanations of the RF/transmission
subsystem and the transport and service multiplex subsystems.
Copies of the standard can be obtained from the NRSC at
http://www.nrscstandards.ora/standards.asp. iBiquity's HD Radio
technology is an implementation of the NRSC-5A IBOC standard.
Further information regarding HD Radio technology can be found at
www.hdradio.com and www.ibiquity.com.
[0008] The HD Radio system includes a radio link subsystem that is
designed primarily for data transmission. It would be desirable to
utilize the radio link subsystem to transmit streaming data such as
audio and video.
SUMMARY OF THE INVENTION
[0009] In one aspect, the invention provided an apparatus for
controlling a data rate in a data client for a digital audio
broadcasting system including a buffer for storing data, a codec
for coding data, and a control module for controlling a bit rate of
the codec in response to a level of the data in the buffer.
[0010] In another aspect, the invention provides a method for
controlling a data rate in a digital audio broadcasting system
including: storing data in a buffer, using a codec to code the data
read from the buffer, and controlling a bit rate of the codec in
response to a level of the data in the buffer.
[0011] In another aspect, the invention provides a method for
controlling a data rate in a digital audio broadcasting system
including: storing data in a buffer, compressing the data, and
outputting compressed data at a rate controlled in response to a
level of data in the buffer.
BRIEF DESCRIPTION OF THE DRAWINGS
[0012] FIG. 1 is a block diagram of a transmitter for use in an
in-band on-channel digital radio broadcasting system.
[0013] FIG. 2 is a schematic representation of a hybrid FM IBOC
waveform.
[0014] FIG. 3 is a schematic representation of an extended hybrid
FM IBOC waveform.
[0015] FIG. 4 is a schematic representation of an all-digital FM
IBOC waveform.
[0016] FIG. 5 is a schematic representation of a hybrid AM IBOC DAB
waveform.
[0017] FIG. 6 is a schematic representation of an all-digital AM
IBOC DAB waveform.
[0018] FIG. 7 is a functional block diagram of an AM IBOC DAB
receiver.
[0019] FIG. 8 is a functional block diagram of an FM IBOC DAB
receiver.
[0020] FIGS. 9a and 9b are diagrams of an IBOC DAB logical protocol
stack from the broadcast perspective.
[0021] FIG. 10 is a diagram of an IBOC DAB logical protocol stack
from the receiver perspective.
[0022] FIG. 11 is a block diagram that illustrates buffering in
accordance with an aspect of the invention.
[0023] FIG. 12 is a pictorial representation of a buffer level.
[0024] FIG. 13 is a schematic representation of interactions
between a data client, an importer, and an exporter.
DETAILED DESCRIPTION OF THE INVENTION
[0025] FIGS. 1-10 and the accompanying description herein provide a
general description of an IBOC system, including broadcasting
equipment structure and operation, receiver structure and
operation, and the structure of IBOC DAB waveforms. FIGS. 11-13 and
the accompanying description herein provide a description of
aspects of the present invention.
IBOC System and Waveforms
[0026] Referring to the drawings, FIG. 1 is a functional block
diagram of the relevant components of a studio site 10, an FM
transmitter site 12, and a studio transmitter link (STL) 14 that
can be used to broadcast an FM IBOC DAB signal. The studio site
includes, among other things, studio automation equipment 34, an
Ensemble Operations Center (EOC) 16 that includes an importer 18,
an exporter 20, an exciter auxiliary service unit (EASU) 22, and an
STL transmitter 48. The transmitter site includes an STL receiver
54, a digital exciter 56 that includes an exciter engine (exgine)
subsystem 58, and an analog exciter 60. While in FIG. 1 the
exporter is resident at a radio station's studio site and the
exciter is located at the transmission site, these elements may be
co-located at the transmission site.
[0027] At the studio site, the studio automation equipment supplies
main program service (MPS) audio 42 to the EASU, MPS data 40 to the
exporter, supplemental program service (SPS) audio 38 to the
importer, and SPS data 36 to the importer. MPS audio serves as the
main audio programming source. In hybrid modes, it preserves the
existing analog radio programming formats in both the analog and
digital transmissions. MPS data, also known as program service data
(PSD), includes information such as music title, artist, album
name, etc. Supplemental program service can include supplementary
audio content as well as program associated data.
[0028] The importer contains hardware and software for supplying
advanced application services (AAS). A "service" is content that is
delivered to users via an IBOC DAB broadcast, and AAS can include
any type of data that is not classified as MPS, SPS, or Station
Information Service (SIS). SIS provides station information, such
as call sign, absolute time, position correlated to GPS, etc.
Examples of AAS data include real-time traffic and weather
information, navigation map updates or other images, electronic
program guides, multimedia programming, other audio services, and
other content. The content for AAS can be supplied by service
providers 44, which provide service data 46 to the importer via an
application program interface (API). The service providers may be a
broadcaster located at the studio site or externally sourced
third-party providers of services and content. The importer can
establish session connections between multiple service providers.
The importer encodes and multiplexes service data 46, SPS audio 38,
and SPS data 36 to produce exporter link data 24, which is output
to the exporter via a data link.
[0029] The exporter 20 contains the hardware and software necessary
to supply the main program service and SIS for broadcasting. The
exporter accepts digital MPS audio 26 over an audio interface and
compresses the audio. The exporter also multiplexes MPS data 40,
exporter link data 24, and the compressed digital MPS audio to
produce exciter link data 52. In addition, the exporter accepts
analog MPS audio 28 over its audio interface and applies a
pre-programmed delay to it to produce a delayed analog MPS audio
signal 30. This analog audio can be broadcast as a backup channel
for hybrid IBOC DAB broadcasts. The delay compensates for the
system delay of the digital MPS audio, allowing receivers to blend
between the digital and analog program without a shift in time. In
an AM transmission system, the delayed MPS audio signal 30 is
converted by the exporter to a mono signal and sent directly to the
STL as part of the exciter link data 52.
[0030] The EASU 22 accepts MPS audio 42 from the studio automation
equipment, rate converts it to the proper system clock, and outputs
two copies of the signal, one digital (26) and one analog (28). The
EASU includes a GPS receiver that is connected to an antenna 25.
The GPS receiver allows the EASU to derive a master clock signal,
which is synchronized to the exciter's clock by use of GPS units.
The EASU provides the master system clock used by the exporter. The
EASU is also used to bypass (or redirect) the analog MPS audio from
being passed through the exporter in the event the exporter has a
catastrophic fault and is no longer operational. The bypassed audio
32 can be fed directly into the STL transmitter, eliminating a
dead-air event.
[0031] STL transmitter 48 receives delayed analog MPS audio 50 and
exciter link data 52. It outputs exciter link data and delayed
analog MPS audio over STL link 14, which may be either
unidirectional or bidirectional. The STL link may be a digital
microwave or Ethernet link, for example, and may use the standard
User Datagram Protocol or the standard TCP/IP.
[0032] The transmitter site includes an STL receiver 54, an exciter
56 and an analog exciter 60. The STL receiver 54 receives exciter
link data, including audio and data signals as well as command and
control messages, over the STL link 14. The exciter link data is
passed to the exciter 56, which produces the IBOC DAB waveform. The
exciter includes a host processor, digital up-converter, RF
up-converter, and exgine subsystem 58. The exgine accepts exciter
link data and modulates the digital portion of the IBOC DAB
waveform. The digital up-converter of exciter 56 converts from
digital-to-analog the baseband portion of the exgine output. The
digital-to-analog conversion is based on a GPS clock, common to
that of the exporter's GPS-based clock derived from the EASU. Thus,
the exciter 56 includes a GPS unit and antenna 57. An alternative
method for synchronizing the exporter and exciter clocks can be
found in U.S. patent application Ser. No. 11/081,267 (Publication
No. 2006/0209941 A1), the disclosure of which is hereby
incorporated by reference. The RF up-converter of the exciter
up-converts the analog signal to the proper in-band channel
frequency. The up-converted signal is then passed to the high power
amplifier 62 and antenna 64 for broadcast. In an AM transmission
system, the exgine subsystem coherently adds the backup analog MPS
audio to the digital waveform in the hybrid mode; thus, the AM
transmission system does not include the analog exciter 60. In
addition, the exciter 56 produces phase and magnitude information
and the analog signal is output directly to the high power
amplifier.
[0033] IBOC DAB signals can be transmitted in both AM and FM radio
bands, using a variety of waveforms. The waveforms include an FM
hybrid IBOC DAB waveform, an FM all-digital IBOC DAB waveform, an
AM hybrid IBOC DAB waveform, and an AM all-digital IBOC DAB
waveform.
[0034] FIG. 2 is a schematic representation of a hybrid FM IBOC
waveform 70. The waveform includes an analog modulated signal 72
located in the center of a broadcast channel 74, a first plurality
of evenly spaced orthogonally frequency division multiplexed
subcarriers 76 in an upper sideband 78, and a second plurality of
evenly spaced orthogonally frequency division multiplexed
subcarriers 80 in a lower sideband 82. The digitally modulated
subcarriers are divided into partitions and various subcarriers are
designated as reference subcarriers. A frequency partition is a
group of 19 OFDM subcarriers containing 18 data subcarriers and one
reference subcarrier.
[0035] The hybrid waveform includes an analog FM-modulated signal,
plus digitally modulated primary main subcarriers. The subcarriers
are located at evenly spaced frequency locations. The subcarrier
locations are numbered from -546 to +546. In the waveform of FIG.
2, the subcarriers are at locations +356 to +546 and -356 to -546.
Each primary main sideband is comprised of ten frequency
partitions. Subcarriers 546 and -546, also included in the primary
main sidebands, are additional reference subcarriers. The amplitude
of each subcarrier can be scaled by an amplitude scale factor.
[0036] FIG. 3 is a schematic representation of an extended hybrid
FM IBOC waveform 90. The extended hybrid waveform is created by
adding primary extended sidebands 92, 94 to the primary main
sidebands present in the hybrid waveform. One, two, or four
frequency partitions can be added to the inner edge of each primary
main sideband. The extended hybrid waveform includes the analog FM
signal plus digitally modulated primary main subcarriers
(subcarriers +356 to +546 and -356 to -546) and some or all primary
extended subcarriers (subcarriers +280 to +355 and -280 to
-355).
[0037] The upper primary extended sidebands include subcarriers 337
through 355 (one frequency partition), 318 through 355 (two
frequency partitions), or 280 through 355 (four frequency
partitions). The lower primary extended sidebands include
subcarriers -337 through -355 (one frequency partition), -318
through -355 (two frequency partitions), or -280 through -355 (four
frequency partitions). The amplitude of each subcarrier can be
scaled by an amplitude scale factor.
[0038] FIG. 4 is a schematic representation of an all-digital FM
IBOC waveform 100. The all-digital waveform is constructed by
disabling the analog signal, fully expanding the bandwidth of the
primary digital sidebands 102, 104, and adding lower-power
secondary sidebands 106, 108 in the spectrum vacated by the analog
signal. The all-digital waveform in the illustrated embodiment
includes digitally modulated subcarriers at subcarrier locations
-546 to +546, without an analog FM signal.
[0039] In addition to the ten main frequency partitions, all four
extended frequency partitions are present in each primary sideband
of the all-digital waveform. Each secondary sideband also has ten
secondary main (SM) and four secondary extended (SX) frequency
partitions. Unlike the primary sidebands, however, the secondary
main frequency partitions are mapped nearer to the channel center
with the extended frequency partitions farther from the center.
[0040] Each secondary sideband also supports a small secondary
protected (SP) region 110, 112 including 12 OFDM subcarriers and
reference subcarriers 279 and -279. The sidebands are referred to
as "protected" because they are located in the area of spectrum
least likely to be affected by analog or digital interference. An
additional reference subcarrier is placed at the center of the
channel (0). Frequency partition ordering of the SP region does not
apply since the SP region does not contain frequency
partitions.
[0041] Each secondary main sideband spans subcarriers 1 through 190
or -1 through -190. The upper secondary extended sideband includes
subcarriers 191 through 266, and the upper secondary protected
sideband includes subcarriers 267 through 278, plus additional
reference subcarrier 279. The lower secondary extended sideband
includes subcarriers -191 through -266, and the lower secondary
protected sideband includes subcarriers -267 through -278, plus
additional reference subcarrier -279. The total frequency span of
the entire all-digital spectrum is 396,803 Hz. The amplitude of
each subcarrier can be scaled by an amplitude scale factor. The
secondary sideband amplitude scale factors can be user selectable.
Any one of the four may be selected for application to the
secondary sidebands.
[0042] In each of the waveforms, the digital signal is modulated
using orthogonal frequency division multiplexing (OFDM). OFDM is a
parallel modulation scheme in which the data stream modulates a
large number of orthogonal subcarriers, which are transmitted
simultaneously. OFDM is inherently flexible, readily allowing the
mapping of logical channels to different groups of subcarriers.
[0043] In the hybrid waveform, the digital signal is transmitted in
primary main (PM) sidebands on either side of the analog FM signal
in the hybrid waveform. The power level of each sideband is
appreciably below the total power in the analog FM signal. The
analog signal may be monophonic or stereo, and may include
subsidiary communications authorization (SCA) channels.
[0044] In the extended hybrid waveform, the bandwidth of the hybrid
sidebands can be extended toward the analog FM signal to increase
digital capacity. This additional spectrum, allocated to the inner
edge of each primary main sideband, is termed the primary extended
(PX) sideband.
[0045] In the all-digital waveform, the analog signal is removed
and the bandwidth of the primary digital sidebands is fully
extended as in the extended hybrid waveform. In addition, this
waveform allows lower-power digital secondary sidebands to be
transmitted in the spectrum vacated by the analog FM signal.
[0046] FIG. 5 is a schematic representation of an AM hybrid IBOC
DAB waveform 120. The hybrid format includes the conventional AM
analog signal 122 (bandlimited to about .+-.5 kHz) along with a
nearly 30 kHz wide DAB signal 124. The spectrum is contained within
a channel 126 having a bandwidth of about 30 kHz. The channel is
divided into upper 130 and lower 132 frequency bands. The upper
band extends from the center frequency of the channel to about +15
kHz from the center frequency. The lower band extends from the
center frequency to about -15 kHz from the center frequency.
[0047] The AM hybrid IBOC DAB signal format in one example
comprises the analog modulated carrier signal 134 plus OFDM
subcarrier locations spanning the upper and lower bands. Coded
digital information representative of the audio or data signals to
be transmitted (program material), is transmitted on the
subcarriers. The symbol rate is less than the subcarrier spacing
due to a guard time between symbols.
[0048] As shown in FIG. 5, the upper band is divided into a primary
section 136, a secondary section 138, and a tertiary section 144.
The lower band is divided into a primary section 140, a secondary
section 142, and a tertiary section 143. For the purpose of this
explanation, the tertiary sections 143 and 144 can be considered to
include a plurality of groups of subcarriers labeled 146, 148, 150
and 152 in FIG. 5. Subcarriers within the tertiary sections that
are positioned near the center of the channel are referred to as
inner subcarriers, and subcarriers within the tertiary sections
that are positioned farther from the center of the channel are
referred to as outer subcarriers. In this example, the power level
of the inner subcarriers in groups 148 and 150 is shown to decrease
linearly with frequency spacing from the center frequency. The
remaining groups of subcarriers 146 and 152 in the tertiary
sections have substantially constant power levels. FIG. 5 also
shows two reference subcarriers 154 and 156 for system control,
whose levels are fixed at a value that is different from the other
sidebands.
[0049] The power of subcarriers in the digital sidebands is
significantly below the total power in the analog AM signal. The
level of each OFDM subcarrier within a given primary or secondary
section is fixed at a constant value. Primary or secondary sections
may be scaled relative to each other. In addition, status and
control information is transmitted on reference subcarriers located
on either side of the main carrier. A separate logical channel,
such as an IBOC Data Service (IDS) channel can be transmitted in
individual subcarriers just above and below the frequency edges of
the upper and lower secondary sidebands. The power level of each
primary OFDM subcarrier is fixed relative to the unmodulated main
analog carrier. However, the power level of the secondary
subcarriers, logical channel subcarriers, and tertiary subcarriers
is adjustable.
[0050] Using the modulation format of FIG. 5, the analog modulated
carrier and the digitally modulated subcarriers are transmitted
within the channel mask specified for standard AM broadcasting in
the United States. The hybrid system uses the analog AM signal for
tuning and backup.
[0051] FIG. 6 is a schematic representation of the subcarrier
assignments for an all-digital AM IBOC DAB waveform. The
all-digital AM IBOC DAB signal 160 includes first and second groups
162 and 164 of evenly spaced subcarriers, referred to as the
primary subcarriers, that are positioned in upper and lower bands
166 and 168. Third and fourth groups 170 and 172 of subcarriers,
referred to as secondary and tertiary subcarriers respectively, are
also positioned in upper and lower bands 166 and 168. Two reference
subcarriers 174 and 176 of the third group lie closest to the
center of the channel. Subcarriers 178 and 180 can be used to
transmit program information data.
[0052] FIG. 7 is a simplified functional block diagram of an AM
IBOC DAB receiver 200. The receiver includes an input 202 connected
to an antenna 204, a tuner or front end 206, and a digital down
converter 208 for producing a baseband signal on line 210. An
analog demodulator 212 demodulates the analog modulated portion of
the baseband signal to produce an analog audio signal on line 214.
A digital demodulator 216 demodulates the digitally modulated
portion of the baseband signal. Then the digital signal is
deinterleaved by a deinterleaver 218, and decoded by a Viterbi
decoder 220. A service demultiplexer 222 separates main and
supplemental program signals from data signals. A processor 224
processes the program signals to produce a digital audio signal on
line 226. The analog and main digital audio signals are blended as
shown in block 228, or a supplemental digital audio signal is
passed through, to produce an audio output on line 230. A data
processor 232 processes the data signals and produces data output
signals on lines 234, 236 and 238. The data signals can include,
for example, a station information service (SIS), main program
service data (MPSD), supplemental program service data (SPSD), and
one or more auxiliary application services (AAS).
[0053] FIG. 8 is a simplified functional block diagram of an FM
IBOC DAB receiver 250. The receiver includes an input 252 connected
to an antenna 254 and a tuner or front end 256. A received signal
is provided to an analog-to-digital converter and digital down
converter 258 to produce a baseband signal at output 260 comprising
a series of complex signal samples. The signal samples are complex
in that each sample comprises a "real" component and an "imaginary"
component, which is sampled in quadrature to the real component. An
analog demodulator 262 demodulates the analog modulated portion of
the baseband signal to produce an analog audio signal on line 264.
The digitally modulated portion of the sampled baseband signal is
next filtered by sideband isolation filter 266, which has a
pass-band frequency response comprising the collective set of
subcarriers f.sub.1-f.sub.n present in the received OFDM signal.
Filter 268 suppresses the effects of a first-adjacent interferer.
Complex signal 298 is routed to the input of acquisition module
296, which acquires or recovers OFDM symbol timing offset or error
and carrier frequency offset or error from the received OFDM
symbols as represented in received complex signal 298. Acquisition
module 296 develops a symbol timing offset .DELTA.t and carrier
frequency offset .DELTA.f, as well as status and control
information. The signal is then demodulated (block 272) to
demodulate the digitally modulated portion of the baseband signal.
Then the digital signal is deinterleaved by a deinterleaver 274,
and decoded by a Viterbi decoder 276. A service demultiplexer 278
separates main and supplemental program signals from data signals.
A processor 280 processes the main and supplemental program signals
to produce a digital audio signal on line 282. The analog and main
digital audio signals are blended as shown in block 284, or the
supplemental program signal is passed through, to produce an audio
output on line 286. A data processor 288 processes the data signals
and produces data output signals on lines 290, 292 and 294. The
data signals can include, for example, a station information
service (SIS), main program service data (MPSD), supplemental
program service data (SPSD), and one or more advanced application
services (AAS).
[0054] In practice, many of the signal processing functions shown
in the receivers of FIGS. 7 and 8 can be implemented using one or
more integrated circuits.
[0055] FIGS. 9a and 9b are diagrams of an IBOC DAB logical protocol
stack from the transmitter perspective. From the receiver
perspective, the logical stack will be traversed in the opposite
direction. Most of the data being passed between the various
entities within the protocol stack are in the form of protocol data
units (PDUs). A PDU is a structured data block that is produced by
a specific layer (or process within a layer) of the protocol stack.
The PDUs of a given layer may encapsulate PDUs from the next higher
layer of the stack and/or include content data and protocol control
information originating in the layer (or process) itself. The PDUs
generated by each layer (or process) in the transmitter protocol
stack are inputs to a corresponding layer (or process) in the
receiver protocol stack.
[0056] As shown in FIGS. 9a and 9b, there is a configuration
administrator 330, which is a system function that supplies
configuration and control information to the various entities
within the protocol stack. The configuration/control information
can include user defined settings, as well as information generated
from within the system such as GPS time and position. The service
interfaces 331 represent the interfaces for all services except
SIS. The service interface may be different for each of the various
types of services. For example, for MPS audio and SPS audio, the
service interface may be an audio card. For MPS data and SPS data
the interfaces may be in the form of different application program
interfaces (APIs). For all other data services the interface is in
the form of a single API. An audio codec 332 encodes both MPS audio
and SPS audio to produce core (Stream 0) and optional enhancement
(Stream 1) streams of MPS and SPS audio encoded packets, which are
passed to audio transport 333. Audio codec 332 also relays unused
capacity status to other parts of the system, thus allowing the
inclusion of opportunistic data. MPS and SPS data is processed by
program service data (PSD) transport 334 to produce MPS and SPS
data PDUs, which are passed to audio transport 333. Audio transport
333 receives encoded audio packets and PSD PDUs and outputs bit
streams containing both compressed audio and program service data.
The SIS transport 335 receives SIS data from the configuration
administrator and generates SIS PDUs. A SIS PDU can contain station
identification and location information, program type, as well as
absolute time and position correlated to GPS. The AAS data
transport 336 receives AAS data from the service interface, as well
as opportunistic bandwidth data from the audio transport, and
generates AAS data PDUs, which can be based on quality of service
parameters. The transport and encoding functions are collectively
referred to as Layer 4 of the protocol stack and the corresponding
transport PDUs are referred to as Layer 4 PDUs or LA PDUs. Layer 2,
which is the channel multiplex layer, (337) receives transport PDUs
from the SIS transport, AAS data transport, and audio transport,
and formats them into Layer 2 PDUs. A Layer 2 PDU includes protocol
control information and a payload, which can be audio, data, or a
combination of audio and data. Layer 2 PDUs are routed through the
correct logical channels to Layer 1 (338), wherein a logical
channel is a signal path that conducts L1 PDUs through Layer 1 with
a specified grade of service. There are multiple Layer 1 logical
channels based on service mode, wherein a service mode is a
specific configuration of operating parameters specifying
throughput, performance level, and selected logical channels. The
number of active Layer 1 logical channels and the characteristics
defining them vary for each service mode. Status information is
also passed between Layer 2 and Layer 1. Layer 1 converts the PDUs
from Layer 2 and system control information into an AM or FM IBOC
DAB waveform for transmission. Layer 1 processing can include
scrambling, channel encoding, interleaving, OFDM subcarrier
mapping, and OFDM signal generation. The output of OFDM signal
generation is a complex, baseband, time domain pulse representing
the digital portion of an IBOC signal for a particular symbol.
Discrete symbols are concatenated to form a continuous time domain
waveform, which is modulated to create an IBOC waveform for
transmission.
[0057] FIG. 10 shows the logical protocol stack from the receiver
perspective. An IBOC waveform is received by the physical layer,
Layer 1 (560), which demodulates the signal and processes it to
separate the signal into logical channels. The number and kind of
logical channels will depend on the service mode, and may include
logical channels P1-P3, PIDS, S1-S5, and SIDS. Layer 1 produces L1
PDUs corresponding to the logical channels and sends the PDUs to
Layer 2 (565), which demultiplexes the L1 PDUs to produce SIS PDUs,
AAS PDUs, PSD PDUs for the main program service and any
supplemental program services, and Stream 0 (core) audio PDUs and
Stream 1 (optional enhanced) audio PDUs. The SIS PDUs are then
processed by the SIS transport 570 to produce SIS data, the AAS
PDUs are processed by the AAS transport 575 to produce AAS data,
and the PSD PDUs are processed by the PSD transport 580 to produce
MPS data (MPSD) and any SPS data (SPSD). The SIS data, AAS data,
MPSD and SPSD are then sent to a user interface 590. The SIS data,
if requested by a user, can then be displayed. Likewise, MPSD,
SPSD, and any text based or graphical AAS data can be displayed.
The Stream 0 and Stream 1 PDUs are processed by Layer 4, comprised
of audio transport 590 and audio decoder 595. There may be up to N
audio transports corresponding to the number of programs received
on the IBOC waveform. Each audio transport produces encoded MPS
packets or SPS packets, corresponding to each of the received
programs. Layer 4 receives control information from the user
interface, including commands such as to store or play programs,
and to seek or scan for radio stations broadcasting an all-digital
or hybrid IBOC signal. Layer 4 also provides status information to
the user interface.
Streaming
[0058] The HD Radio system uses an importer to collect all advanced
data services and secondary program services into a package that is
delivered to the exporter/exciter. As shown in and described with
respect to FIG. 1, the main program service audio is multiplexed in
at the exporter. These services are then put on various logical
channels which map to sets of OFDM carriers. Clients, who may be
service providers, for example, connect to the importer and then
follow a request/respond mechanism initiated by the importer. It is
at this point where the secondary audio or data services may be
clocked at a different fundamental rate than the asking side, i.e.,
the importer.
[0059] The HD Radio system services two types of data, streaming
(e.g., audio, video) and all other kinds of data. In this regard,
data in the form of an audio stream does not refer to the main or
supplemental program services, but rather to other audio that may
be sent via the data transport of the HD Radio broadcast system,
called the Radio Link Service (RLS). Unlike other types of data, a
stream has time requirements for delivery. If these time
requirements are not met, there will most likely be gaps in the
user's reception of the service where there is no data (possibly
audio) to play. RLS previously has not been used to stream data
(audio or other) because there is a non-deterministic bandwidth
allocation algorithm at its core, which uses a partial High-Level
Data Link Control (HDLC) framing technique. This HDLC framing may
insert escape characters into the data stream, which is dependent
on the data content, thus not easily lending itself to
determinism.
[0060] In one aspect, the invention attempts to treat the
non-deterministic behavior of the bandwidth allocation algorithm of
RLS as a clock mismatch issue. If the generating clock (transmit
side) differs in frequency enough from the reconstruction clock
(receiving side), a possible data overflow or underflow would occur
at the receiver. This situation can be addressed on the transmit
side according to an aspect of the present invention by tying a
buffer management policy to the codec rate, thus allowing the
buffer level to drive a slow control loop that will in turn drive
the codec rate. This will have the effect of mitigating any
non-deterministic behavior in the partial HDLC framing technique
employed by the RLS. This requires that the codec be bit rate
controllable and phase continuous at the switches. Changes in
frequency are phase continuous when they do not cause
discontinuities in the phase (or amplitude) of the output
signal.
[0061] FIG. 11 is a block diagram of a Client/Importer/Exporter
Connection, showing the flow of data requests and responses between
the Importer and Exporter. In this example, a data source 700
supplies data (e.g. audio, video, or other data) to a processor
702, which can be, for example an audio card, video card, or other
processor to produce a data signal on line 704 that is supplied to
a data client 706. The timing of the processor is controlled by a
clock 708. The data client includes a buffer 710, a buffer
management module 712, a codec 714, and control module 716. The
data client sends data to the importer 718 on line 720 and receives
data requests on line 722. The importer in turn determines which
clients need to supply data based on a service-to-logical channel
mapping. A plurality of clients can be connected to the importer.
The exporter 724 receives data in the form of protocol data units
on line 726, and sends a PDU indication signal to the importer on
line 728. The exporter requests data at the exciter clock rate, and
drives the importer output rate and the rate of data requests from
the importer to clients. The clients may be clocked at various
rates which may or may not be tied to the exciter clock. The
exporter uses the PDUs to produce an output signal on line 730.
[0062] The data on line 704 can be in the form of pulse code
modulated (PCM) audio samples, which are stored in a buffer 710.
The buffer management module 712 receives data requests on line
722, retrieves data from the buffer, and sends the retrieved data
to the codec 714 on line 732. The control module 716 monitors the
level of data stored in the buffer and controls the operation of
the codec in response to the level of data in the buffer. The codec
outputs compressed data on line 720. The buffer management module
712, buffer 710, and control module 716 can be implemented as
software components in the data client. Alternatively, these
functions can be implemented as software components in the
importer.
[0063] The buffer management module 712 would typically use some
locking mechanism such as a semaphore to insure that the "state" of
the buffer is preserved. The state of the buffer is simply the
level or the amount of data in the buffer. The locking semaphore
would be taken if an output action is requested and released after
the state of the buffer is updated to reflect less data in the
buffer. In a similar fashion, the locking semaphore would be taken
by the input before a new data packet is committed to the buffer,
and released after it is committed and the state is updated. In
this way the buffer manager ensures that the asynchronous nature of
the inputting to the buffer and the outputting from the buffer
preserve a consistent buffer state. The state of the buffer will be
important to assess whether or not any external action is required
to keep the level or state of the buffer essentially constant.
[0064] From the data client side, any deviation from an exact
requesting rate will appear to be a clock mismatch, even if it is
the result of extra bytes put in the data stream by the RLS
transport protocol. Thus, even if the client and exporter/exciter
clocks are matched, the non-deterministic behavior of the partial
HDLC framing will introduce what will appear to be clock frequency
mismatches or phase drifts. In one aspect, the method of buffer
management of the present invention will mitigate the clock
mismatch issue regardless of how it has been introduced.
[0065] The control module 716 may employ a simple slow data control
loop to determine if any action is required to increase or decrease
the buffer level. The single observable parameter is the buffer
level (or state). FIG. 12 is a pictorial illustration of an example
buffer level, showing how the level may evolve over time. The
dashed lines 740 and 742 represent boundary levels of the buffer.
Whenever the buffer level equals or exceeds these levels, either
greater than the overflow boundary 740 or less than the underflow
boundary 742 levels, an action is taken, for example at time
t.sub.1, t.sub.2 and t.sub.3. This action would increase or
decrease the compression rate of the codec 710, so that the level
would stay in the acceptable range of level in between the dashed
lines.
[0066] Thus, in one example, the dashed lines 740 and 742 represent
first and second predetermined buffer threshold levels, wherein the
first threshold level is higher than the second threshold level. If
the buffer level exceeds the first threshold level, then the buffer
level is considered to be too high. If the buffer level is below
the second threshold level, then the buffer level is considered to
be too low. If the buffer level is between the two threshold
levels, then the buffer level is considered to be acceptable.
[0067] If the level is above the first threshold level, a command
can be sent by the control module 716 to the codec 714 to reduce
the bit-rate; thus the codec will produce less data per block of
PCM audio data. If the buffer level is below the second threshold
level, the control module can either simply hold off the data
requests from the importer, thus forfeiting some of the bandwidth
it has been allocated, or it can increase the bit-rate of the codec
so the codec produces more data per block of data samples.
[0068] The bit rate control would depend on what is available from
the codec being used. In order to use this type of control system,
the codec would have to be phase continuous at the bit rate
changes; otherwise there may be artifacts or discontinuities in the
output waveform. However, if the codec could switch bit rate phase
continuously, then when an action is deemed necessary by the
control module a switch to the next highest or lowest codec rate
would take place. This approach assumes that the initial rate of
samples output by the codec is very close to bandwidth allocation
rates of the data channel. A large mismatch in these parameters may
result in a buffer overflow or underflow condition that is too
large for correction by this technique.
[0069] As previously described, the underlying reason for the codec
rate adjustment may be either too many added escape bytes, or a
fundamental clock rate mismatch. Regardless of the reason for
making an adjustment, this technique will work well within bounds.
Generally, the buffer should be big enough to hold data so that a
trend can be discerned. This is a result of the burstyness of the
data transfer in and out of the buffer. Thus, the adjustment in the
codec output rate should be kept small so that clear trends can be
ascertained. These basic provisos are standard issues when
designing any control loop.
[0070] FIG. 13 is a schematic representation of the exchange of
messages and data among the data client, the importer, and the
exporter. The exchange is initiated by the exporter sending a "PDU
Needed" indication to the importer, thus requesting that the
importer send it data. The importer also receives a Log-In and
Setup Session communication from the data client, establishing a
dialogue that will allow the data client to send data to the
importer. A buffer is then established and filled within the data
client. Thereafter, the data client responds to a Get Data Request
from the importer by reading the appropriate size packets that have
been generated and placed in the buffer and sending the packets to
the importer (the Get Data Response). The importer in turn formats
the data into a PDU that is sent to the exporter as a PDU Response.
This process of generating PDU Responses continues for as long as
the data client has data to be broadcast. In order to mitigate
effects from mismatched clocks and the non-exact bandwidth of RLS
within the HD Radio data channel, the control module can be
employed to increase the codec rate if the buffer begins to
deplete, or decrease the codec rate if the buffer begins to build
up. This control loop allows the data application to continue
without interruption and without regard for exact clock matching or
non-deterministic RLS bandwidth effects, so long as the bit-rate of
the codec can be switched phase continuously.
[0071] While the present invention has been described in terms of
its preferred embodiment, it will be understood by those skilled in
the art that various modifications can be made to the described
embodiments without departing from the scope of the invention as
set forth in the claims.
* * * * *
References